From stephen at picardogroup.com  Wed Apr  1 12:13:33 2020
From: stephen at picardogroup.com (stephen at picardogroup.com)
Date: Wed, 01 Apr 2020 05:13:33 -0700
Subject: [Freeswitch-users] SIP help
Message-ID: <20200401051333.0e1bd4d5c5064b420440751b21b10e46.8a309fa8a5.wbe@email13.godaddy.com>

An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200401/647f5866/attachment.html>

From david.villasmil.work at gmail.com  Wed Apr  1 15:09:23 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 1 Apr 2020 16:09:23 +0100
Subject: [Freeswitch-users] SIP help
In-Reply-To: <20200401051333.0e1bd4d5c5064b420440751b21b10e46.8a309fa8a5.wbe@email13.godaddy.com>
References: <20200401051333.0e1bd4d5c5064b420440751b21b10e46.8a309fa8a5.wbe@email13.godaddy.com>
Message-ID: <CAFGRPVp-G3WSTnTBkfr5NAzO7Zn8nuct9s370FeRRaVw+NdVAQ@mail.gmail.com>

If you remove that # what happens?

On Wed, 1 Apr 2020 at 14:00, <stephen at picardogroup.com> wrote:

> Using Flowroute for all traffic with FS 1.6  Pop we use will be retired in
> June. Need to change.
>
> Our test string does not work and Flowroute support has no suggestions!
>
> Is is the test string we send and the response, can anyone point out what
> is wrong?
>
> originate
> {jitterbuffer_msec=180,ignore_early_media=true,originate_timeout=30}sofia/gateway/flowroute/Acct#
> *1xxxxxxxxxx at us-east-nj.sip.flowroute.com &park()
>
> 2020-03-28 15:13:13.763325 [NOTICE] switch_channel.c:1104 New Channel
> sofia/external/Acct#*1xxxxxxxxxx at us-east-nj.sip.flowroute.com
> [de80ab1b-744a-4ff1-a299-32f2a01f05ad]
> 2020-03-28 15:13:43.003328 [NOTICE] switch_ivr_originate.c:3612 Hangup
> sofia/external/Acct#*1xxxxxxxxxx at us-east-nj.sip.flowroute.com
> [CS_CONSUME_MEDIA] [NO_ANSWER]
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200401/096b7dcb/attachment.html>

From brian at freeswitch.com  Wed Apr  1 22:32:39 2020
From: brian at freeswitch.com (Brian West)
Date: Wed, 1 Apr 2020 17:32:39 -0500
Subject: [Freeswitch-users] SIP help
In-Reply-To: <CAFGRPVp-G3WSTnTBkfr5NAzO7Zn8nuct9s370FeRRaVw+NdVAQ@mail.gmail.com>
References: <20200401051333.0e1bd4d5c5064b420440751b21b10e46.8a309fa8a5.wbe@email13.godaddy.com>
 <CAFGRPVp-G3WSTnTBkfr5NAzO7Zn8nuct9s370FeRRaVw+NdVAQ@mail.gmail.com>
Message-ID: <CAEJMVkD5WXRuJ_hfNFSCE29=czQ5Td2-kgWUNycUOWzMmLtX9Q@mail.gmail.com>

It's probably URL encoding it, look at your sip packet.   a '#' is not
allowed in the user portion of a sip URI.

/b


On Wed, Apr 1, 2020 at 11:29 AM David Villasmil <
david.villasmil.work at gmail.com> wrote:

> If you remove that # what happens?
>
> On Wed, 1 Apr 2020 at 14:00, <stephen at picardogroup.com> wrote:
>
>> Using Flowroute for all traffic with FS 1.6  Pop we use will be retired
>> in June. Need to change.
>>
>> Our test string does not work and Flowroute support has no suggestions!
>>
>> Is is the test string we send and the response, can anyone point out what
>> is wrong?
>>
>> originate
>> {jitterbuffer_msec=180,ignore_early_media=true,originate_timeout=30}sofia/gateway/flowroute/Acct#
>> *1xxxxxxxxxx at us-east-nj.sip.flowroute.com &park()
>>
>> 2020-03-28 15:13:13.763325 [NOTICE] switch_channel.c:1104 New Channel
>> sofia/external/Acct#*1xxxxxxxxxx at us-east-nj.sip.flowroute.com
>> [de80ab1b-744a-4ff1-a299-32f2a01f05ad]
>> 2020-03-28 15:13:43.003328 [NOTICE] switch_ivr_originate.c:3612 Hangup
>> sofia/external/Acct#*1xxxxxxxxxx at us-east-nj.sip.flowroute.com
>> [CS_CONSUME_MEDIA] [NO_ANSWER]
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 

Brian West | Co-founder and Developer

Need Commercial support? email sales at freeswitch.com

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

Email: brian at freeswitch.com

Mobile: 918-424-9378

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

[image: https://www.facebook.com/signalwireinc?src=email]
<https://www.facebook.com/freeswitch> [image:
https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200401/01735d84/attachment.html>

From ahabiba at gmail.com  Thu Apr  2 09:44:24 2020
From: ahabiba at gmail.com (Ahmed habiba)
Date: Thu, 2 Apr 2020 12:44:24 +0300
Subject: [Freeswitch-users] Voice Conference Voice Choppy
Message-ID: <D90DBFD2-FEC6-4E24-B85C-7DC485C356E1@gmail.com>

Hello,

I’m trying to use FreeSWITCH voice conference, however hear some cuts is the voice from time to time some minor distortions, I look into code and I see something not sure if I’m right or wrong, when the conference loop is not able to read from input it sleep for 100 MS? Kindly advice how to overcome this issue.

To Note, I’m using server with "Xeon E3-1245v2 (4c/8th) - 32GB” and I’m using it for 3 to 4 users only, nothing else ring on this machine.

Thanks,
Ahmed Habiba.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200402/f1ba104d/attachment-0001.html>

From gmaruzz at gmail.com  Thu Apr  2 10:23:29 2020
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 2 Apr 2020 12:23:29 +0200
Subject: [Freeswitch-users] Voice Conference Voice Choppy
In-Reply-To: <D90DBFD2-FEC6-4E24-B85C-7DC485C356E1@gmail.com>
References: <D90DBFD2-FEC6-4E24-B85C-7DC485C356E1@gmail.com>
Message-ID: <CALXCt0qhf81CGbQTz-6j_C-oTCK3Oq+tgVa_TRxddkmNYp8Ctg@mail.gmail.com>

If you are running in baremetal (eg, no VMs), then check packet
loss/jitter...



On Thu, Apr 2, 2020 at 11:53 AM Ahmed habiba <ahabiba at gmail.com> wrote:

> Hello,
>
> I’m trying to use FreeSWITCH voice conference, however hear some cuts is
> the voice from time to time some minor distortions, I look into code and I
> see something not sure if I’m right or wrong, when the conference loop is
> not able to read from input it sleep for 100 MS? Kindly advice how to
> overcome this issue.
>
> To Note, I’m using server with "Xeon E3-1245v2 (4c/8th) - 32GB” and I’m
> using it for 3 to 4 users only, nothing else ring on this machine.
>
> Thanks,
> Ahmed Habiba.
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200402/0fe80f93/attachment.html>

From stephen at picardogroup.com  Thu Apr  2 12:48:45 2020
From: stephen at picardogroup.com (stephen at picardogroup.com)
Date: Thu, 02 Apr 2020 05:48:45 -0700
Subject: [Freeswitch-users] PoP Flowroute
Message-ID: <20200402054845.0e1bd4d5c5064b420440751b21b10e46.18f31877dd.wbe@email13.godaddy.com>

An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200402/2bb64e56/attachment.html>

From nathan at robotics.net  Thu Apr  2 13:18:24 2020
From: nathan at robotics.net (Nathan Stratton)
Date: Thu, 2 Apr 2020 09:18:24 -0400
Subject: [Freeswitch-users] iOS push notifications
Message-ID: <CAHubkyPLdLL5h0n4Fu+vGf0otc_T=Qpmp-TbP65e4pOd4FYC+Q@mail.gmail.com>

Does anyone have iOS push notifications working on FreeSWITCH 10 that is
willing to help me get it working for a fee?

><>
nathan stratton
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200402/4b4b82e4/attachment.html>

From gmaruzz at gmail.com  Thu Apr  2 13:38:41 2020
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 2 Apr 2020 15:38:41 +0200
Subject: [Freeswitch-users] iOS push notifications
In-Reply-To: <CAHubkyPLdLL5h0n4Fu+vGf0otc_T=Qpmp-TbP65e4pOd4FYC+Q@mail.gmail.com>
References: <CAHubkyPLdLL5h0n4Fu+vGf0otc_T=Qpmp-TbP65e4pOd4FYC+Q@mail.gmail.com>
Message-ID: <CALXCt0rM5gXjntnZT1oWbjMOWVO_Wz2RDvDT+0Qc59ijOz=zYg@mail.gmail.com>

write a lua script and use google gcm (works for both android and ios)



On Thu, Apr 2, 2020 at 3:35 PM Nathan Stratton <nathan at robotics.net> wrote:

> Does anyone have iOS push notifications working on FreeSWITCH 10 that is
> willing to help me get it working for a fee?
>
> ><>
> nathan stratton
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200402/32ecaab1/attachment.html>

From gmaruzz at gmail.com  Thu Apr  2 13:41:00 2020
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 2 Apr 2020 15:41:00 +0200
Subject: [Freeswitch-users] iOS push notifications
In-Reply-To: <CALXCt0rM5gXjntnZT1oWbjMOWVO_Wz2RDvDT+0Qc59ijOz=zYg@mail.gmail.com>
References: <CAHubkyPLdLL5h0n4Fu+vGf0otc_T=Qpmp-TbP65e4pOd4FYC+Q@mail.gmail.com>
 <CALXCt0rM5gXjntnZT1oWbjMOWVO_Wz2RDvDT+0Qc59ijOz=zYg@mail.gmail.com>
Message-ID: <CALXCt0puLeCjfw3-2msL-uagW4mYJtwkuwsw9bbyeinZDXnq2w@mail.gmail.com>

lua, perl, whatever FreeSWITCH script language (not necessarily lua).



On Thu, Apr 2, 2020 at 3:38 PM Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> write a lua script and use google gcm (works for both android and ios)
>
>
>
> On Thu, Apr 2, 2020 at 3:35 PM Nathan Stratton <nathan at robotics.net>
> wrote:
>
>> Does anyone have iOS push notifications working on FreeSWITCH 10 that is
>> willing to help me get it working for a fee?
>>
>> ><>
>> nathan stratton
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
>

-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200402/660a4faa/attachment.html>

From ahabiba at gmail.com  Thu Apr  2 14:04:27 2020
From: ahabiba at gmail.com (Ahmed habiba)
Date: Thu, 2 Apr 2020 17:04:27 +0300
Subject: [Freeswitch-users] Voice Conference Voice Choppy(Ahabiba)
In-Reply-To: <mailman.0.1585828802.75136.freeswitch-users@lists.freeswitch.org>
References: <mailman.0.1585828802.75136.freeswitch-users@lists.freeswitch.org>
Message-ID: <0E12DA3A-4701-4330-B7A0-EBFF40B0A7C3@gmail.com>

Hi,

Here is the Jitter calculated from my machine against the server using perf - the server is baremetal 

[ ID] Interval           Transfer     Bitrate         Jitter    Lost/Total Datagrams
[  5]   0.00-20.00  sec  2.38 MBytes  1.00 Mbits/sec  0.000 ms  0/1786 (0%)  sender
[  5]   0.00-20.00  sec  2.38 MBytes  1000 Kbits/sec  0.321 ms  1/1786 (0.056%)  receiver

Perf result from server side:
[ ID] Interval           Transfer     Bandwidth       Jitter    Lost/Total Datagrams
[  5]   0.00-20.09  sec  2.38 MBytes   996 Kbits/sec  0.321 ms  1/1786 (0.056%)  


Client iperf command:
 iperf3 -u -c 176.31.123.52 -b 1M -i 1 -t 20 


Thanks,
Ahmed Habiba.

> 
>   1. Re: Voice Conference Voice Choppy (Giovanni Maruzzelli)
> 
> From: Giovanni Maruzzelli <gmaruzz at gmail.com>
> Subject: Re: [Freeswitch-users] Voice Conference Voice Choppy
> Date: 2 April 2020 at 1:23:29 PM GMT+3
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Reply-To: gmaruzz at opentelecom.it
> 
> 
> If you are running in baremetal (eg, no VMs), then check packet loss/jitter...
> 
> 
> 
> On Thu, Apr 2, 2020 at 11:53 AM Ahmed habiba <ahabiba at gmail.com <mailto:ahabiba at gmail.com>> wrote:
> Hello,
> 
> I’m trying to use FreeSWITCH voice conference, however hear some cuts is the voice from time to time some minor distortions, I look into code and I see something not sure if I’m right or wrong, when the conference loop is not able to read from input it sleep for 100 MS? Kindly advice how to overcome this issue.
> 
> To Note, I’m using server with "Xeon E3-1245v2 (4c/8th) - 32GB” and I’m using it for 3 to 4 users only, nothing else ring on this machine.
> 
> Thanks,
> Ahmed Habiba.
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/>
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community <https://signalwire.community/>
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com <mailto:sales at freeswitch.com>
> https://freeswitch.com <https://freeswitch.com/>
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss <https://freeswitch.com/oss>
> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
> https://cluecon.com <https://cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> https://freeswitch.com <https://freeswitch.com/>
> 
> -- 
> Sincerely,
> 
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
> 
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200402/63470b75/attachment-0001.html>

From ahabiba at gmail.com  Fri Apr  3 16:58:22 2020
From: ahabiba at gmail.com (Ahmed habiba)
Date: Fri, 3 Apr 2020 19:58:22 +0300
Subject: [Freeswitch-users] Voice Conference Voice
In-Reply-To: <mailman.2128.1585836302.1226.freeswitch-users@lists.freeswitch.org>
References: <mailman.2128.1585836302.1226.freeswitch-users@lists.freeswitch.org>
Message-ID: <B9800D1C-D7EC-4699-9FAB-B8A25ED6F43D@gmail.com>

Hello,

Any one can help?

"I look into code and I see something not sure if I’m right or wrong, when the conference loop is not able to read from input it sleep for 100 MS”?

Thanks,
Ahmed Habiba.

> 
> 
> From: Ahmed habiba <ahabiba at gmail.com>
> Subject: Re: [Freeswitch-users] Voice Conference Voice Choppy(Ahabiba)
> Date: 2 April 2020 at 5:04:27 PM GMT+3
> To: freeswitch-users at lists.freeswitch.org
> 
> 
> Hi,
> 
> Here is the Jitter calculated from my machine against the server using perf - the server is baremetal 
> 
> [ ID] Interval           Transfer     Bitrate         Jitter    Lost/Total Datagrams
> [  5]   0.00-20.00  sec  2.38 MBytes  1.00 Mbits/sec  0.000 ms  0/1786 (0%)  sender
> [  5]   0.00-20.00  sec  2.38 MBytes  1000 Kbits/sec  0.321 ms  1/1786 (0.056%)  receiver
> 
> Perf result from server side:
> [ ID] Interval           Transfer     Bandwidth       Jitter    Lost/Total Datagrams
> [  5]   0.00-20.09  sec  2.38 MBytes   996 Kbits/sec  0.321 ms  1/1786 (0.056%)  
> 
> 
> Client iperf command:
>  iperf3 -u -c 176.31.123.52 -b 1M -i 1 -t 20 
> 
> 
> Thanks,
> Ahmed Habiba.
> 
>> 
>>   1. Re: Voice Conference Voice Choppy (Giovanni Maruzzelli)
>> 
>> From: Giovanni Maruzzelli <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>>
>> Subject: Re: [Freeswitch-users] Voice Conference Voice Choppy
>> Date: 2 April 2020 at 1:23:29 PM GMT+3
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
>> Reply-To: gmaruzz at opentelecom.it <mailto:gmaruzz at opentelecom.it>
>> 
>> 
>> If you are running in baremetal (eg, no VMs), then check packet loss/jitter...
>> 
>> 
>> 
>> On Thu, Apr 2, 2020 at 11:53 AM Ahmed habiba <ahabiba at gmail.com <mailto:ahabiba at gmail.com>> wrote:
>> Hello,
>> 
>> I’m trying to use FreeSWITCH voice conference, however hear some cuts is the voice from time to time some minor distortions, I look into code and I see something not sure if I’m right or wrong, when the conference loop is not able to read from input it sleep for 100 MS? Kindly advice how to overcome this issue.
>> 
>> To Note, I’m using server with "Xeon E3-1245v2 (4c/8th) - 32GB” and I’m using it for 3 to 4 users only, nothing else ring on this machine.
>> 
>> Thanks,
>> Ahmed Habiba.
>> _________________________________________________________________________
>> 
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/>
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>> Build your next product on our scalable cloud platform.
>> 
>> Join our online community to chat in real time https://signalwire.community <https://signalwire.community/>
>> 
>> Professional FreeSWITCH Services
>> sales at freeswitch.com <mailto:sales at freeswitch.com>
>> https://freeswitch.com <https://freeswitch.com/>
>> 
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss <https://freeswitch.com/oss>
>> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
>> https://cluecon.com <https://cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> https://freeswitch.com <https://freeswitch.com/>
>> 
>> -- 
>> Sincerely,
>> 
>> Giovanni Maruzzelli
>> OpenTelecom.IT
>> cell: +39 347 266 56 18
>> 
>> 
> 
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200403/348ddcc3/attachment.html>

From gmaruzz at gmail.com  Fri Apr  3 19:47:52 2020
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 3 Apr 2020 21:47:52 +0200
Subject: [Freeswitch-users] Voice Conference Voice
In-Reply-To: <B9800D1C-D7EC-4699-9FAB-B8A25ED6F43D@gmail.com>
References: <mailman.2128.1585836302.1226.freeswitch-users@lists.freeswitch.org>
 <B9800D1C-D7EC-4699-9FAB-B8A25ED6F43D@gmail.com>
Message-ID: <CALXCt0qCXg9w1XcLZNYkKoDer7uKgD79Qot2FX6c=153LjrqtQ@mail.gmail.com>

Ahmed,

I would tell you: restart from scratch, format the hard disk, start from a
Debian 10 64 bit Server  minimal installation on bare machine, then follow
strictly and exactly the instruction on confluence for Debian 10 (buster)
install from packages

https://freeswitch.org/confluence/display/FREESWITCH/Debian+10+Buster

Do not change nothing. Just cut and paste.

Then test again

The conference code is OK, million people is using it.

Have a nice weekend,

-giovanni



On Fri, Apr 3, 2020 at 7:31 PM Ahmed habiba <ahabiba at gmail.com> wrote:

> Hello,
>
> Any one can help?
>
> "I look into code and I see something not sure if I’m right or wrong, when
> the conference loop is not able to read from input it sleep for 100 MS”?
>
> Thanks,
> Ahmed Habiba.
>
>
>
> *From: *Ahmed habiba <ahabiba at gmail.com>
> *Subject: **Re: [Freeswitch-users] Voice Conference Voice Choppy(Ahabiba)*
> *Date: *2 April 2020 at 5:04:27 PM GMT+3
> *To: *freeswitch-users at lists.freeswitch.org
>
>
> Hi,
>
> Here is the Jitter calculated from my machine against the server using
> perf - the server is baremetal
>
> [ ID] Interval           Transfer     Bitrate         Jitter    Lost/Total
> Datagrams
> [  5]   0.00-20.00  sec  2.38 MBytes  1.00 Mbits/sec  0.000 ms  0/1786
> (0%)  sender
> [  5]   0.00-20.00  sec  2.38 MBytes  1000 Kbits/sec  0.321 ms  1/1786
> (0.056%)  receiver
>
> Perf result from server side:
> [ ID] Interval           Transfer     Bandwidth       Jitter    Lost/Total
> Datagrams
> [  5]   0.00-20.09  sec  2.38 MBytes   996 Kbits/sec  0.321 ms  1/1786
> (0.056%)
>
>
> Client iperf command:
>  iperf3 -u -c 176.31.123.52 -b 1M -i 1 -t 20
>
>
> Thanks,
> Ahmed Habiba.
>
>
>   1. Re: Voice Conference Voice Choppy (Giovanni Maruzzelli)
>
> *From: *Giovanni Maruzzelli <gmaruzz at gmail.com>
> *Subject: **Re: [Freeswitch-users] Voice Conference Voice Choppy*
> *Date: *2 April 2020 at 1:23:29 PM GMT+3
> *To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Reply-To: *gmaruzz at opentelecom.it
>
>
> If you are running in baremetal (eg, no VMs), then check packet
> loss/jitter...
>
>
>
> On Thu, Apr 2, 2020 at 11:53 AM Ahmed habiba <ahabiba at gmail.com> wrote:
>
>> Hello,
>>
>> I’m trying to use FreeSWITCH voice conference, however hear some cuts is
>> the voice from time to time some minor distortions, I look into code and I
>> see something not sure if I’m right or wrong, when the conference loop is
>> not able to read from input it sleep for 100 MS? Kindly advice how to
>> overcome this issue.
>>
>> To Note, I’m using server with "Xeon E3-1245v2 (4c/8th) - 32GB” and I’m
>> using it for 3 to 4 users only, nothing else ring on this machine.
>>
>> Thanks,
>> Ahmed Habiba.
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
>
>
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200403/3c3ca99c/attachment-0001.html>

From david.villasmil.work at gmail.com  Fri Apr  3 20:56:24 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 3 Apr 2020 21:56:24 +0100
Subject: [Freeswitch-users] Voice Conference Voice
In-Reply-To: <CALXCt0qCXg9w1XcLZNYkKoDer7uKgD79Qot2FX6c=153LjrqtQ@mail.gmail.com>
References: <mailman.2128.1585836302.1226.freeswitch-users@lists.freeswitch.org>
 <B9800D1C-D7EC-4699-9FAB-B8A25ED6F43D@gmail.com>
 <CALXCt0qCXg9w1XcLZNYkKoDer7uKgD79Qot2FX6c=153LjrqtQ@mail.gmail.com>
Message-ID: <CAFGRPVr-zixFVi0cuGwQ-dj24kyG9a=u27OBZCexQ52y8rZgKw@mail.gmail.com>

And make sure it’s not your network, usually the culprit.

On Fri, 3 Apr 2020 at 21:05, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:

> Ahmed,
>
> I would tell you: restart from scratch, format the hard disk, start from a
> Debian 10 64 bit Server  minimal installation on bare machine, then follow
> strictly and exactly the instruction on confluence for Debian 10 (buster)
> install from packages
>
> https://freeswitch.org/confluence/display/FREESWITCH/Debian+10+Buster
>
> Do not change nothing. Just cut and paste.
>
> Then test again
>
> The conference code is OK, million people is using it.
>
> Have a nice weekend,
>
> -giovanni
>
>
>
> On Fri, Apr 3, 2020 at 7:31 PM Ahmed habiba <ahabiba at gmail.com> wrote:
>
>> Hello,
>>
>> Any one can help?
>>
>> "I look into code and I see something not sure if I’m right or wrong,
>> when the conference loop is not able to read from input it sleep for 100
>> MS”?
>>
>> Thanks,
>> Ahmed Habiba.
>>
>>
>>
>> *From: *Ahmed habiba <ahabiba at gmail.com>
>> *Subject: **Re: [Freeswitch-users] Voice Conference Voice
>> Choppy(Ahabiba)*
>> *Date: *2 April 2020 at 5:04:27 PM GMT+3
>> *To: *freeswitch-users at lists.freeswitch.org
>>
>>
>> Hi,
>>
>> Here is the Jitter calculated from my machine against the server using
>> perf - the server is baremetal
>>
>> [ ID] Interval           Transfer     Bitrate         Jitter
>> Lost/Total Datagrams
>> [  5]   0.00-20.00  sec  2.38 MBytes  1.00 Mbits/sec  0.000 ms  0/1786
>> (0%)  sender
>> [  5]   0.00-20.00  sec  2.38 MBytes  1000 Kbits/sec  0.321 ms  1/1786
>> (0.056%)  receiver
>>
>> Perf result from server side:
>> [ ID] Interval           Transfer     Bandwidth       Jitter
>> Lost/Total Datagrams
>> [  5]   0.00-20.09  sec  2.38 MBytes   996 Kbits/sec  0.321 ms  1/1786
>> (0.056%)
>>
>>
>> Client iperf command:
>>  iperf3 -u -c 176.31.123.52 -b 1M -i 1 -t 20
>>
>>
>> Thanks,
>> Ahmed Habiba.
>>
>>
>>   1. Re: Voice Conference Voice Choppy (Giovanni Maruzzelli)
>>
>> *From: *Giovanni Maruzzelli <gmaruzz at gmail.com>
>> *Subject: **Re: [Freeswitch-users] Voice Conference Voice Choppy*
>> *Date: *2 April 2020 at 1:23:29 PM GMT+3
>> *To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Reply-To: *gmaruzz at opentelecom.it
>>
>>
>> If you are running in baremetal (eg, no VMs), then check packet
>> loss/jitter...
>>
>>
>>
>> On Thu, Apr 2, 2020 at 11:53 AM Ahmed habiba <ahabiba at gmail.com> wrote:
>>
>>> Hello,
>>>
>>> I’m trying to use FreeSWITCH voice conference, however hear some cuts is
>>> the voice from time to time some minor distortions, I look into code and I
>>> see something not sure if I’m right or wrong, when the conference loop is
>>> not able to read from input it sleep for 100 MS? Kindly advice how to
>>> overcome this issue.
>>>
>>> To Note, I’m using server with "Xeon E3-1245v2 (4c/8th) - 32GB” and I’m
>>> using it for 3 to 4 users only, nothing else ring on this machine.
>>>
>>> Thanks,
>>> Ahmed Habiba.
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> OpenTelecom.IT
>> cell: +39 347 266 56 18
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200403/a14d4e5c/attachment.html>

From victor.chukalovskiy at gmail.com  Sat Apr  4 03:45:49 2020
From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy)
Date: Fri, 3 Apr 2020 23:45:49 -0400
Subject: [Freeswitch-users] How to connect SIP call to rtsp stream
Message-ID: <3bec69bd-5bd1-7338-f48a-83f5fa8af068@gmail.com>

Good day, I'm trying connect incoming video call into rtsp stream (e.g. 
security camera h.264 video only or h.264 video + PCMU audio). I tried 
these two approaches:

<action application="playback" 
data="av://rtsp://192.168.2.7/streaming/channels/103"/>
or
<action application="playback" 
data="rtsp://192.168.2.7/streaming/channels/103"/>

And FS doesn't like it:

[ERR] switch_core_file.c:306 Invalid file format [av] for 
[rtsp://192.168.2.7/streaming/channels/103]!
or
[ERR] switch_core_file.c:306 Invalid file format [rtsp] for 
[192.168.2.7/streaming/channels/103]!

What is the right syntax to accomplish this? Also, which modules need to 
be compiled & loaded for this to work?

Many thanks,
-victor


From imfanee at gmail.com  Sat Apr  4 12:43:43 2020
From: imfanee at gmail.com (Faisal Hanif)
Date: Sat, 4 Apr 2020 17:43:43 +0500
Subject: [Freeswitch-users] How to connect SIP call to rtsp stream
In-Reply-To: <3bec69bd-5bd1-7338-f48a-83f5fa8af068@gmail.com>
References: <3bec69bd-5bd1-7338-f48a-83f5fa8af068@gmail.com>
Message-ID: <CAO5BruBp8Xa62yErwRW6o41kL0i2WCirjQQ4zbw6PVyKNC_Atw@mail.gmail.com>

you probably missing
mod_localstream
mod_h26x
mod_rtc

On Sat, 4 Apr 2020 at 09:37, Victor Chukalovskiy <
victor.chukalovskiy at gmail.com> wrote:

> Good day, I'm trying connect incoming video call into rtsp stream (e.g.
> security camera h.264 video only or h.264 video + PCMU audio). I tried
> these two approaches:
>
> <action application="playback"
> data="av://rtsp://192.168.2.7/streaming/channels/103"/>
> or
> <action application="playback"
> data="rtsp://192.168.2.7/streaming/channels/103"/>
>
> And FS doesn't like it:
>
> [ERR] switch_core_file.c:306 Invalid file format [av] for
> [rtsp://192.168.2.7/streaming/channels/103]!
> or
> [ERR] switch_core_file.c:306 Invalid file format [rtsp] for
> [192.168.2.7/streaming/channels/103]!
>
> What is the right syntax to accomplish this? Also, which modules need to
> be compiled & loaded for this to work?
>
> Many thanks,
> -victor
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Regards,

Faisal Hanif
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200404/e444e02b/attachment.html>

From girish.dharmaraj at gmail.com  Sat Apr  4 14:31:45 2020
From: girish.dharmaraj at gmail.com (Giri)
Date: Sat, 4 Apr 2020 22:31:45 +0800
Subject: [Freeswitch-users] SIP 302 Redirect
Message-ID: <CALv+5v7oKEdo9W1iw3icMgbasZ=nNzbXE4TMVgrm5yOYu3X8UA@mail.gmail.com>

Good Day, I have a scenario where the Outbound GW sends SIP 302 and all I
need is to send the SIP 302 message to the Inbound.  No Routing needs to
take place, is this possible?

With Best Regards,
Girish Dharmaraj
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200404/658fca4b/attachment.html>

From loidang at hoiio.com  Sat Apr  4 15:35:58 2020
From: loidang at hoiio.com (Loi Dang)
Date: Sat, 4 Apr 2020 22:35:58 +0700
Subject: [Freeswitch-users] SIP 302 Redirect
In-Reply-To: <CALv+5v7oKEdo9W1iw3icMgbasZ=nNzbXE4TMVgrm5yOYu3X8UA@mail.gmail.com>
References: <CALv+5v7oKEdo9W1iw3icMgbasZ=nNzbXE4TMVgrm5yOYu3X8UA@mail.gmail.com>
Message-ID: <CADMnEC9ada3K-j1pUa9gCShwq7sX8mKf20KPq6R93fiVDb5=Lg@mail.gmail.com>

Hi, try manual-redirect option in SIP profile?
rgds,
Loi Dang

On Sat, Apr 4, 2020, 10:12 PM Giri <girish.dharmaraj at gmail.com> wrote:

> Good Day, I have a scenario where the Outbound GW sends SIP 302 and all I
> need is to send the SIP 302 message to the Inbound.  No Routing needs to
> take place, is this possible?
>
> With Best Regards,
> Girish Dharmaraj
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200404/4035a5d0/attachment.html>

From victor.chukalovskiy at gmail.com  Sat Apr  4 15:45:21 2020
From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy)
Date: Sat, 4 Apr 2020 11:45:21 -0400
Subject: [Freeswitch-users] How to connect SIP call to rtsp stream
In-Reply-To: <CAO5BruBp8Xa62yErwRW6o41kL0i2WCirjQQ4zbw6PVyKNC_Atw@mail.gmail.com>
References: <3bec69bd-5bd1-7338-f48a-83f5fa8af068@gmail.com>
 <CAO5BruBp8Xa62yErwRW6o41kL0i2WCirjQQ4zbw6PVyKNC_Atw@mail.gmail.com>
Message-ID: <59d54ad0-3641-c057-4d2e-4a8b2dcf6cc2@gmail.com>

Hmm, I have mod_local_stream & mod_h26x already. I now compiled & loaded 
mod_rtc But having exact same problem:

[ERR] switch_core_file.c:306 Invalid file format [av] for 
[rtsp://192.168.2.7/streaming/channels/103 
<http://192.168.2.7/streaming/channels/103>]!

Wonder if anything else I should try?



On 2020-04-04 8:43 a.m., Faisal Hanif wrote:
> you probably missing
> mod_localstream
> mod_h26x
> mod_rtc
>
> On Sat, 4 Apr 2020 at 09:37, Victor Chukalovskiy 
> <victor.chukalovskiy at gmail.com <mailto:victor.chukalovskiy at gmail.com>> 
> wrote:
>
>     Good day, I'm trying connect incoming video call into rtsp stream
>     (e.g.
>     security camera h.264 video only or h.264 video + PCMU audio). I
>     tried
>     these two approaches:
>
>     <action application="playback"
>     data="av://rtsp://192.168.2.7/streaming/channels/103
>     <http://192.168.2.7/streaming/channels/103>"/>
>     or
>     <action application="playback"
>     data="rtsp://192.168.2.7/streaming/channels/103
>     <http://192.168.2.7/streaming/channels/103>"/>
>
>     And FS doesn't like it:
>
>     [ERR] switch_core_file.c:306 Invalid file format [av] for
>     [rtsp://192.168.2.7/streaming/channels/103
>     <http://192.168.2.7/streaming/channels/103>]!
>     or
>     [ERR] switch_core_file.c:306 Invalid file format [rtsp] for
>     [192.168.2.7/streaming/channels/103
>     <http://192.168.2.7/streaming/channels/103>]!
>
>     What is the right syntax to accomplish this? Also, which modules
>     need to
>     be compiled & loaded for this to work?
>
>     Many thanks,
>     -victor
>
>     _________________________________________________________________________
>
>     The FreeSWITCH project is sponsored by SignalWire
>     https://signalwire.com
>     Enhance your FreeSWITCH install with disruptive priced SMS and
>     PSTN services.
>     Build your next product on our scalable cloud platform.
>
>     Join our online community to chat in real time
>     https://signalwire.community
>
>     Professional FreeSWITCH Services
>     sales at freeswitch.com <mailto:sales at freeswitch.com>
>     https://freeswitch.com
>
>     Official FreeSWITCH Sites
>     https://freeswitch.com/oss
>     https://freeswitch.org/confluence
>     https://cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     https://freeswitch.com
>
>
>
> -- 
> Regards,
>
> Faisal Hanif
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200404/7e3ccccf/attachment-0001.html>

From girish.dharmaraj at gmail.com  Sun Apr  5 10:53:13 2020
From: girish.dharmaraj at gmail.com (Giri)
Date: Sun, 5 Apr 2020 18:53:13 +0800
Subject: [Freeswitch-users] SIP 302 Redirect
In-Reply-To: <CADMnEC9ada3K-j1pUa9gCShwq7sX8mKf20KPq6R93fiVDb5=Lg@mail.gmail.com>
References: <CALv+5v7oKEdo9W1iw3icMgbasZ=nNzbXE4TMVgrm5yOYu3X8UA@mail.gmail.com>
 <CADMnEC9ada3K-j1pUa9gCShwq7sX8mKf20KPq6R93fiVDb5=Lg@mail.gmail.com>
Message-ID: <CALv+5v6D7Mk9g3M9XOeY=HhwTpzuDNuL-dvwerYNiZf==RfdKw@mail.gmail.com>

Hello Sir,

I tried that, but it just goes into the redirect context and fails finding
no match and call is released with 503.

On Sat, Apr 4, 2020, 23:36 Loi Dang <loidang at hoiio.com> wrote:

> Hi, try manual-redirect option in SIP profile?
> rgds,
> Loi Dang
>
> On Sat, Apr 4, 2020, 10:12 PM Giri <girish.dharmaraj at gmail.com> wrote:
>
>> Good Day, I have a scenario where the Outbound GW sends SIP 302 and all I
>> need is to send the SIP 302 message to the Inbound.  No Routing needs to
>> take place, is this possible?
>>
>> With Best Regards,
>> Girish Dharmaraj
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200405/22e1d0a5/attachment.html>

From sdevoy at bizfocused.com  Sun Apr  5 22:19:08 2020
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Sun, 5 Apr 2020 22:19:08 +0000
Subject: [Freeswitch-users] MySQL setup
Message-ID: <BL0PR01MB45157BA5A1A3EAFA9DCE2885BBC50@BL0PR01MB4515.prod.exchangelabs.com>

Hi,

I am running fs 1.10 on Debian 10 64bit.

I have searched for help configuring FreeSwitch to use MySQL.  I have configured db.conf.xml for Mysql:
<param name="core-db-dsn" value="odbc://DRIVER=mysql;SERVER=serveriphere;UID=userhere;PWD=passwordhere;DATABASE=freeswitch;OPTION=67108864" />

I have reloaded, restarted and rebooted. FreeSwitch continues to use sqlite.

Is there more to configure?
Do I need to install unixODBC and/or MySql driver?

Here is the error:
2020-04-05 20:57:48.159620 [ERR] switch_odbc.c:368 STATE: 01000 CODE 0 ERROR: [unixODBC][Driver Manager]Can't open lib 'mysql' : file not found

2020-04-05 20:57:48.159658 [CRIT] switch_core_sqldb.c:646 Failure to connect to ODBC DRIVER=mysql;SERVER=127.0.0.1;UID=xxxxxx;PWD=XXXXXXXXXX;DATABASE=freeswitch;OPTION=67108864!
2020-04-05 20:57:48.160130 [CONSOLE] switch_loadable_module.c:1804 Successfully Loaded [mod_db]
2020-04-05 20:57:48.160146 [NOTICE] switch_loadable_module.c:350 Adding Application 'db'
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200405/8a85310b/attachment.html>

From dujinfang at gmail.com  Mon Apr  6 00:45:10 2020
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 6 Apr 2020 08:45:10 +0800
Subject: [Freeswitch-users] How to connect SIP call to rtsp stream
In-Reply-To: <59d54ad0-3641-c057-4d2e-4a8b2dcf6cc2@gmail.com>
References: <3bec69bd-5bd1-7338-f48a-83f5fa8af068@gmail.com>
 <CAO5BruBp8Xa62yErwRW6o41kL0i2WCirjQQ4zbw6PVyKNC_Atw@mail.gmail.com>
 <59d54ad0-3641-c057-4d2e-4a8b2dcf6cc2@gmail.com>
Message-ID: <CAJs+-5oOOekatndH9u-zwCLJF4=ZhfLqWWD2xDKEYps8Da8QzQ@mail.gmail.com>

load mod_av

On Sun, Apr 5, 2020 at 12:19 AM Victor Chukalovskiy <
victor.chukalovskiy at gmail.com> wrote:

> Hmm, I have mod_local_stream & mod_h26x already. I now compiled & loaded
> mod_rtc But having exact same problem:
>
> [ERR] switch_core_file.c:306 Invalid file format [av] for [rtsp://
> 192.168.2.7/streaming/channels/103]!
>
> Wonder if anything else I should try?
>
>
>
> On 2020-04-04 8:43 a.m., Faisal Hanif wrote:
>
> you probably missing
> mod_localstream
> mod_h26x
> mod_rtc
>
> On Sat, 4 Apr 2020 at 09:37, Victor Chukalovskiy <
> victor.chukalovskiy at gmail.com> wrote:
>
>> Good day, I'm trying connect incoming video call into rtsp stream (e.g.
>> security camera h.264 video only or h.264 video + PCMU audio). I tried
>> these two approaches:
>>
>> <action application="playback"
>> data="av://rtsp://192.168.2.7/streaming/channels/103"/>
>> or
>> <action application="playback"
>> data="rtsp://192.168.2.7/streaming/channels/103"/>
>>
>> And FS doesn't like it:
>>
>> [ERR] switch_core_file.c:306 Invalid file format [av] for
>> [rtsp://192.168.2.7/streaming/channels/103]!
>> or
>> [ERR] switch_core_file.c:306 Invalid file format [rtsp] for
>> [192.168.2.7/streaming/channels/103]!
>>
>> What is the right syntax to accomplish this? Also, which modules need to
>> be compiled & loaded for this to work?
>>
>> Many thanks,
>> -victor
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
> Regards,
>
> Faisal Hanif
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time https://signalwire.community
>
> Professional FreeSWITCH Servicessales at freeswitch.comhttps://freeswitch.com
>
> Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200406/3c64bd87/attachment.html>

From dujinfang at gmail.com  Mon Apr  6 00:50:18 2020
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 6 Apr 2020 08:50:18 +0800
Subject: [Freeswitch-users] Mod_conference with many participants.
 Question for experts
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAAAyq5Nx/9DRDrZ3hLNfSk2jCgAAAEAAAAH1u+aSboX1HjV4zgUKc+EcBAAAAAA==@Sauvaire.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAAAyq5Nx/9DRDrZ3hLNfSk2jCgAAAEAAAAH1u+aSboX1HjV4zgUKc+EcBAAAAAA==@Sauvaire.com>
Message-ID: <CAJs+-5rT64wgkzitkk6AnUuEpLMaGy271_LsWgapC8Q--1vNkw@mail.gmail.com>

You provided a lot of details but missed some important points. What OS?
What version of FreeSWITCH?

I have been running 600+ members across 5 servers even some with videos in
one conference and not have any issue. Most members were muted.

On Tue, Mar 31, 2020 at 4:25 AM Gilles SAUVAIRE <Gilles at sauvaire.com> wrote:

> Hello,
>
> I am making a conference with 100 participants, on an 8-core server, 16
> gigabytes of ram dedicated to my freeswitch.
> 98 participant are in mute mode and only two participant active.
>
> 8 times out of 10 the conference does not work. (very poor quality hatched
> audio)
> 2 times out of 10 the conference works for a few minutes then becomes
> unstable. (very poor quality hatched audio)
>
> A htop at that time shows that all the cores are used but not more than
> 30%.
> There is also plenty of memory available on the server.
>
>     <profile name = "000">
>
>       <param name = "domain" value = "$$ {domain}" />
>       <param name = "rate" value = "8000" />
>       <param name = "interval" value = "20" />
>       <param name = "caller-controls" value = "none" />
>       <param name = "moderator-controls" value = "none" />
>       <param name = "moh-sound" value = "silence" />
>       <param name = "suppress-events" value = "start-talking, stop-talking"
> />
>
>       <! - Energy level required for audio to be sent to the other users ->
>       <param name = "energy-level" value = "100" />
>       <param name = "comfort-noise" value = "0" />
>       <param name = "conference-flags" value = "audio-always" />
>
>     </profile>
>
> I have already found a solution, just put interval at 60.
>       <param name = "interval" value = "60" />
> With this the conference with 100 participants and only 2 active works.
>
> I have many qustions
>
> 1)
> It's strange, the conference with all the members in mute attribute,
> shouldn't it normally do nothing?
> Mute members should be transparent when it comes to mixing, is that just
> broadcast?
> ( conference data = 66123 at 000++flags{endconf|nomoh|mute} )
> Did I make a mistake in my profile? Is it normal ?
>
>
> 2)
> Is there an easier way to broadcast?
> That is, a call that "talks" to 90 calls?
> My idea is to make a conference with 2 or 3 participants and to broadcast
> this conference to all the other participants since he is not speaking.
>
> many thanks to you for your advice and ideas.
>
> Gilles
>
>
>
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200406/0d6204ed/attachment-0001.html>

From loidang at hoiio.com  Mon Apr  6 02:50:15 2020
From: loidang at hoiio.com (Loi Dang)
Date: Mon, 6 Apr 2020 09:50:15 +0700
Subject: [Freeswitch-users] SIP 302 Redirect
In-Reply-To: <CALv+5v6D7Mk9g3M9XOeY=HhwTpzuDNuL-dvwerYNiZf==RfdKw@mail.gmail.com>
References: <CALv+5v7oKEdo9W1iw3icMgbasZ=nNzbXE4TMVgrm5yOYu3X8UA@mail.gmail.com>
 <CADMnEC9ada3K-j1pUa9gCShwq7sX8mKf20KPq6R93fiVDb5=Lg@mail.gmail.com>
 <CALv+5v6D7Mk9g3M9XOeY=HhwTpzuDNuL-dvwerYNiZf==RfdKw@mail.gmail.com>
Message-ID: <CADMnEC8YuZc103wKB8JtHNVNW_AmkEnhaiyiP8PteHHPkppteQ@mail.gmail.com>

try * wildcard matching in your `redirected` context's extension, then do a
<action application="redirect" data="sip:${destination_number}@$${sip_ip}"/>
in the end.
Hope this helps.

rgds,
Loi Dang

On Sun, Apr 5, 2020 at 6:26 PM Giri <girish.dharmaraj at gmail.com> wrote:

> Hello Sir,
>
> I tried that, but it just goes into the redirect context and fails finding
> no match and call is released with 503.
>
> On Sat, Apr 4, 2020, 23:36 Loi Dang <loidang at hoiio.com> wrote:
>
>> Hi, try manual-redirect option in SIP profile?
>> rgds,
>> Loi Dang
>>
>> On Sat, Apr 4, 2020, 10:12 PM Giri <girish.dharmaraj at gmail.com> wrote:
>>
>>> Good Day, I have a scenario where the Outbound GW sends SIP 302 and all
>>> I need is to send the SIP 302 message to the Inbound.  No Routing needs to
>>> take place, is this possible?
>>>
>>> With Best Regards,
>>> Girish Dharmaraj
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200406/fb73e41b/attachment.html>

From david.villasmil.work at gmail.com  Mon Apr  6 03:08:15 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 6 Apr 2020 03:08:15 +0000
Subject: [Freeswitch-users] MySQL setup
In-Reply-To: <BL0PR01MB45157BA5A1A3EAFA9DCE2885BBC50@BL0PR01MB4515.prod.exchangelabs.com>
References: <BL0PR01MB45157BA5A1A3EAFA9DCE2885BBC50@BL0PR01MB4515.prod.exchangelabs.com>
Message-ID: <CAFGRPVrLGVEGT8KUbKc2TC5q2PFYL6R0NqBfU+zFyuUb3e+=kg@mail.gmail.com>

does i_sql work? Have you installed libmysql client?

Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337


On Sun, Apr 5, 2020 at 11:24 PM Sean Devoy <sdevoy at bizfocused.com> wrote:

> Hi,
>
>
>
> I am running fs 1.10 on Debian 10 64bit.
>
>
>
> I have searched for help configuring FreeSwitch to use MySQL.  I have
> configured db.conf.xml for Mysql:
>
> <param name="core-db-dsn"
> value="odbc://DRIVER=mysql;SERVER=serveriphere;UID=userhere;PWD=passwordhere;DATABASE=freeswitch;OPTION=67108864"
> />
>
>
>
> I have reloaded, restarted and rebooted. FreeSwitch continues to use
> sqlite.
>
>
>
> Is there more to configure?
>
> Do I need to install unixODBC and/or MySql driver?
>
>
>
> Here is the error:
>
> 2020-04-05 20:57:48.159620 [ERR] switch_odbc.c:368 STATE: 01000 CODE 0
> ERROR: [unixODBC][Driver Manager]Can't open lib 'mysql' : file not found
>
>
>
> 2020-04-05 20:57:48.159658 [CRIT] switch_core_sqldb.c:646 Failure to
> connect to ODBC
> DRIVER=mysql;SERVER=127.0.0.1;UID=xxxxxx;PWD=XXXXXXXXXX;DATABASE=freeswitch;OPTION=67108864!
>
> 2020-04-05 20:57:48.160130 [CONSOLE] switch_loadable_module.c:1804
> Successfully Loaded [mod_db]
>
> 2020-04-05 20:57:48.160146 [NOTICE] switch_loadable_module.c:350 Adding
> Application 'db'
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200406/dc6f7754/attachment.html>

From vldtoma at gmail.com  Mon Apr  6 09:22:35 2020
From: vldtoma at gmail.com (Vlad Toma)
Date: Mon, 6 Apr 2020 12:22:35 +0300
Subject: [Freeswitch-users] MySQL setup
In-Reply-To: <BL0PR01MB45157BA5A1A3EAFA9DCE2885BBC50@BL0PR01MB4515.prod.exchangelabs.com>
References: <BL0PR01MB45157BA5A1A3EAFA9DCE2885BBC50@BL0PR01MB4515.prod.exchangelabs.com>
Message-ID: <CAFLU3Xg8pvyw4LQFve8rg6gfyAc+Qt_=4NiSCV+Y7=MUWx-9_Q@mail.gmail.com>

Hello,

You can either install both and configure them , or you can use this
directly with mariadb  <param name="core-db-dsn"
value="mariadb://Server=localhost;Database=freeswitch;Uid=freeswitch;Pwd=pass;"
/>

On Mon, Apr 6, 2020 at 1:26 AM Sean Devoy <sdevoy at bizfocused.com> wrote:

> Hi,
>
>
>
> I am running fs 1.10 on Debian 10 64bit.
>
>
>
> I have searched for help configuring FreeSwitch to use MySQL.  I have
> configured db.conf.xml for Mysql:
>
> <param name="core-db-dsn"
> value="odbc://DRIVER=mysql;SERVER=serveriphere;UID=userhere;PWD=passwordhere;DATABASE=freeswitch;OPTION=67108864"
> />
>
>
>
> I have reloaded, restarted and rebooted. FreeSwitch continues to use
> sqlite.
>
>
>
> Is there more to configure?
>
> Do I need to install unixODBC and/or MySql driver?
>
>
>
> Here is the error:
>
> 2020-04-05 20:57:48.159620 [ERR] switch_odbc.c:368 STATE: 01000 CODE 0
> ERROR: [unixODBC][Driver Manager]Can't open lib 'mysql' : file not found
>
>
>
> 2020-04-05 20:57:48.159658 [CRIT] switch_core_sqldb.c:646 Failure to
> connect to ODBC
> DRIVER=mysql;SERVER=127.0.0.1;UID=xxxxxx;PWD=XXXXXXXXXX;DATABASE=freeswitch;OPTION=67108864!
>
> 2020-04-05 20:57:48.160130 [CONSOLE] switch_loadable_module.c:1804
> Successfully Loaded [mod_db]
>
> 2020-04-05 20:57:48.160146 [NOTICE] switch_loadable_module.c:350 Adding
> Application 'db'
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200406/af6b6fd9/attachment-0001.html>

From david.villasmil.work at gmail.com  Mon Apr  6 10:46:09 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 6 Apr 2020 11:46:09 +0100
Subject: [Freeswitch-users] MySQL setup
In-Reply-To: <CAFLU3Xg8pvyw4LQFve8rg6gfyAc+Qt_=4NiSCV+Y7=MUWx-9_Q@mail.gmail.com>
References: <BL0PR01MB45157BA5A1A3EAFA9DCE2885BBC50@BL0PR01MB4515.prod.exchangelabs.com>
 <CAFLU3Xg8pvyw4LQFve8rg6gfyAc+Qt_=4NiSCV+Y7=MUWx-9_Q@mail.gmail.com>
Message-ID: <CAFGRPVpm5y7AFiY5G9o+MyHC7Xz1EiEmdeXM4bK_LMfkgfyecQ@mail.gmail.com>

+1 The best choice

On Mon, 6 Apr 2020 at 11:07, Vlad Toma <vldtoma at gmail.com> wrote:

> Hello,
>
> You can either install both and configure them , or you can use this
> directly with mariadb  <param name="core-db-dsn"
> value="mariadb://Server=localhost;Database=freeswitch;Uid=freeswitch;Pwd=pass;"
> />
>
> On Mon, Apr 6, 2020 at 1:26 AM Sean Devoy <sdevoy at bizfocused.com> wrote:
>
>> Hi,
>>
>>
>>
>> I am running fs 1.10 on Debian 10 64bit.
>>
>>
>>
>> I have searched for help configuring FreeSwitch to use MySQL.  I have
>> configured db.conf.xml for Mysql:
>>
>> <param name="core-db-dsn"
>> value="odbc://DRIVER=mysql;SERVER=serveriphere;UID=userhere;PWD=passwordhere;DATABASE=freeswitch;OPTION=67108864"
>> />
>>
>>
>>
>> I have reloaded, restarted and rebooted. FreeSwitch continues to use
>> sqlite.
>>
>>
>>
>> Is there more to configure?
>>
>> Do I need to install unixODBC and/or MySql driver?
>>
>>
>>
>> Here is the error:
>>
>> 2020-04-05 20:57:48.159620 [ERR] switch_odbc.c:368 STATE: 01000 CODE 0
>> ERROR: [unixODBC][Driver Manager]Can't open lib 'mysql' : file not found
>>
>>
>>
>> 2020-04-05 20:57:48.159658 [CRIT] switch_core_sqldb.c:646 Failure to
>> connect to ODBC
>> DRIVER=mysql;SERVER=127.0.0.1;UID=xxxxxx;PWD=XXXXXXXXXX;DATABASE=freeswitch;OPTION=67108864!
>>
>> 2020-04-05 20:57:48.160130 [CONSOLE] switch_loadable_module.c:1804
>> Successfully Loaded [mod_db]
>>
>> 2020-04-05 20:57:48.160146 [NOTICE] switch_loadable_module.c:350 Adding
>> Application 'db'
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200406/e96015be/attachment.html>

From girish.dharmaraj at gmail.com  Mon Apr  6 10:49:07 2020
From: girish.dharmaraj at gmail.com (Giri)
Date: Mon, 6 Apr 2020 18:49:07 +0800
Subject: [Freeswitch-users] SIP 302 Redirect
In-Reply-To: <CADMnEC8YuZc103wKB8JtHNVNW_AmkEnhaiyiP8PteHHPkppteQ@mail.gmail.com>
References: <CALv+5v7oKEdo9W1iw3icMgbasZ=nNzbXE4TMVgrm5yOYu3X8UA@mail.gmail.com>
 <CADMnEC9ada3K-j1pUa9gCShwq7sX8mKf20KPq6R93fiVDb5=Lg@mail.gmail.com>
 <CALv+5v6D7Mk9g3M9XOeY=HhwTpzuDNuL-dvwerYNiZf==RfdKw@mail.gmail.com>
 <CADMnEC8YuZc103wKB8JtHNVNW_AmkEnhaiyiP8PteHHPkppteQ@mail.gmail.com>
Message-ID: <CALv+5v5+01U_ecGdFdpEN1r+T3L5WigzmxC6G+ZZtZd9q9zhAw@mail.gmail.com>

Hello Sir,

Noted with thanks. Relaly appreciate your response.  I am trying it but I
need another clarification if you can see the below message received from
the outbound on the 302 SIP , I have an header Subject: Callinfo, is it
possible to store this Callinfo to a variable and pass it to the inbound
side as it is without making any changes to the Subject header .

SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 192.168.200.18:5080
;rport=5080;branch=z9hG4bK2Z7c6ZaXSQt5a
   Contact: <sip:11111 at XX.XX.XX.XX>
   To: <sip:1234567895 at 192.168.200.44:30007>;tag=aa14db7c
   From: "104"<sip:FreeSWITCH at 192.168.200.44:30007>;tag=gHg6SeSDN569j
   Call-ID: a80638c1-f296-1238-21ab-3417ebe52801
   CSeq: 18531190 INVITE
   Subject:
callinfo=012020001A01316P100000000000448048898116681921495000035040100100100586400035010100100100586400035020100100100586400031020100100100586400031020100100100586400010
   User-Agent: hmd/2.0
   Content-Length: 0

With Best Regards,
Girish Dharmaraj


On Mon, Apr 6, 2020 at 10:51 AM Loi Dang <loidang at hoiio.com> wrote:

> try * wildcard matching in your `redirected` context's extension, then do
> a <action application="redirect" data="sip:${destination_number}@$${sip_ip}"/>
> in the end.
> Hope this helps.
>
> rgds,
> Loi Dang
>
> On Sun, Apr 5, 2020 at 6:26 PM Giri <girish.dharmaraj at gmail.com> wrote:
>
>> Hello Sir,
>>
>> I tried that, but it just goes into the redirect context and fails
>> finding no match and call is released with 503.
>>
>> On Sat, Apr 4, 2020, 23:36 Loi Dang <loidang at hoiio.com> wrote:
>>
>>> Hi, try manual-redirect option in SIP profile?
>>> rgds,
>>> Loi Dang
>>>
>>> On Sat, Apr 4, 2020, 10:12 PM Giri <girish.dharmaraj at gmail.com> wrote:
>>>
>>>> Good Day, I have a scenario where the Outbound GW sends SIP 302 and all
>>>> I need is to send the SIP 302 message to the Inbound.  No Routing needs to
>>>> take place, is this possible?
>>>>
>>>> With Best Regards,
>>>> Girish Dharmaraj
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200406/0ebf8874/attachment-0001.html>

From jurijs.ivolga at gmail.com  Mon Apr  6 11:16:49 2020
From: jurijs.ivolga at gmail.com (Jurijs Ivolga)
Date: Mon, 6 Apr 2020 14:16:49 +0300
Subject: [Freeswitch-users] MySQL setup
In-Reply-To: <CAFGRPVpm5y7AFiY5G9o+MyHC7Xz1EiEmdeXM4bK_LMfkgfyecQ@mail.gmail.com>
References: <BL0PR01MB45157BA5A1A3EAFA9DCE2885BBC50@BL0PR01MB4515.prod.exchangelabs.com>
 <CAFLU3Xg8pvyw4LQFve8rg6gfyAc+Qt_=4NiSCV+Y7=MUWx-9_Q@mail.gmail.com>
 <CAFGRPVpm5y7AFiY5G9o+MyHC7Xz1EiEmdeXM4bK_LMfkgfyecQ@mail.gmail.com>
Message-ID: <CAOrA2UZZ5UXou5maF=OdgE-UdBcE3nPyWRhC1VpacwFUg6H-uQ@mail.gmail.com>

Hi,

If you go with unixODBC:

1) You need to install unixodbc (apt-get -y install unixodbc)
2) Install odbc connector for MariaDB, for me it worked only 2.0.* version
on Debian 10 (wget
https://downloads.mariadb.com/Connectors/odbc/connector-odbc-2.0.19/mariadb-connector-odbc-2.0.19-ga-debian-x86_64.tar.gz
&& tar -xvzf mariadb-co* && install lib/libmaodbc.so)
3) Configure odbc connector, something like this:
https://mariadb.com/kb/en/creating-a-data-source-with-mariadb-connectorodbc/
4) test if you can connect to MariaDB using isql command.

At this point there will left only Freeswitch to configure. I believe you
do not have installed unixodbc or odbc connector.

Jurijs


On Mon, Apr 6, 2020 at 1:56 PM David Villasmil <
david.villasmil.work at gmail.com> wrote:

> +1 The best choice
>
> On Mon, 6 Apr 2020 at 11:07, Vlad Toma <vldtoma at gmail.com> wrote:
>
>> Hello,
>>
>> You can either install both and configure them , or you can use this
>> directly with mariadb  <param name="core-db-dsn"
>> value="mariadb://Server=localhost;Database=freeswitch;Uid=freeswitch;Pwd=pass;"
>> />
>>
>> On Mon, Apr 6, 2020 at 1:26 AM Sean Devoy <sdevoy at bizfocused.com> wrote:
>>
>>> Hi,
>>>
>>>
>>>
>>> I am running fs 1.10 on Debian 10 64bit.
>>>
>>>
>>>
>>> I have searched for help configuring FreeSwitch to use MySQL.  I have
>>> configured db.conf.xml for Mysql:
>>>
>>> <param name="core-db-dsn"
>>> value="odbc://DRIVER=mysql;SERVER=serveriphere;UID=userhere;PWD=passwordhere;DATABASE=freeswitch;OPTION=67108864"
>>> />
>>>
>>>
>>>
>>> I have reloaded, restarted and rebooted. FreeSwitch continues to use
>>> sqlite.
>>>
>>>
>>>
>>> Is there more to configure?
>>>
>>> Do I need to install unixODBC and/or MySql driver?
>>>
>>>
>>>
>>> Here is the error:
>>>
>>> 2020-04-05 20:57:48.159620 [ERR] switch_odbc.c:368 STATE: 01000 CODE 0
>>> ERROR: [unixODBC][Driver Manager]Can't open lib 'mysql' : file not found
>>>
>>>
>>>
>>> 2020-04-05 20:57:48.159658 [CRIT] switch_core_sqldb.c:646 Failure to
>>> connect to ODBC
>>> DRIVER=mysql;SERVER=127.0.0.1;UID=xxxxxx;PWD=XXXXXXXXXX;DATABASE=freeswitch;OPTION=67108864!
>>>
>>> 2020-04-05 20:57:48.160130 [CONSOLE] switch_loadable_module.c:1804
>>> Successfully Loaded [mod_db]
>>>
>>> 2020-04-05 20:57:48.160146 [NOTICE] switch_loadable_module.c:350 Adding
>>> Application 'db'
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200406/db159757/attachment.html>

From victor.chukalovskiy at gmail.com  Mon Apr  6 22:39:04 2020
From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy)
Date: Mon, 6 Apr 2020 18:39:04 -0400
Subject: [Freeswitch-users] Problems with TLS after upgrading to Buster
In-Reply-To: <20191113200131.GA738@darth.lan>
References: <020001d598de$4fb28980$ef179c80$@behrend-cs.de>
 <20191112202715.GA3584@darth.lan> <20191112211454.GA3609@darth.lan>
 <031d01d599a1$8d30aee0$a7920ca0$@behrend-cs.de>
 <20191113200131.GA738@darth.lan>
Message-ID: <4bb9eb51-8a7d-d32e-629b-da6546d4eb3d@gmail.com>

Old thread, but in case it helps anyone. For SessionTalk + FreeSwitch, 
you need to set openssl.cnf params rock bottom low:

[system_default_sect]
MinProtocol = TLSv1
CipherString = DEFAULT at SECLEVEL=1

I've followed-up with SessionTalk support to check on TLS V1.2 and stronger cipher suite. However my hopes are low since they haven't though of it on their own. App store feedback pending...


On 2019-11-13 3:01 p.m., Sebastian Kemper wrote:
> On Tue, Nov 12, 2019 at 10:38:40PM +0100, Walter Behrend wrote:
>> Btw, I think there is a problem in freeswitch - if for example I
>> configure stunnel, there is no problem with specifying accepting also
>> older TLS standards without the need of changing the MinProtocol
>> setting within the openssl.cnf file. As a user or admin, I would
>> normally expect the tls-version parameter to do the same job for me...
> Hi Walter,
>
> I guess that's a point of view. I was quite happy to find that OpenSSL
> enforces the restrictions set in /etc/ssl/openssl.cnf also when used
> through FreeSWITCH. I'd find it rather strange if it didn't, honestly.
> If they're the default settings then they have to be enforced whenever
> OpenSSL is used, in my opinion.
>
> I've tested with an updated message digest in gentls_cert (SHA256 like
> you suggested) and can confirm it's working properly with this. I've
> sent a pull request via GitHub to FS.
>
> Regards,
> Seb
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



From freeswitch at mikesdriveway.com  Fri Apr  3 13:11:24 2020
From: freeswitch at mikesdriveway.com (MikeKulls)
Date: Sat, 4 Apr 2020 00:11:24 +1100
Subject: [Freeswitch-users] Freeswitch drops calls after 32 seconds
In-Reply-To: <mailman.2146.1585916092.1226.freeswitch-users@lists.freeswitch.org>
References: <mailman.2146.1585916092.1226.freeswitch-users@lists.freeswitch.org>
Message-ID: <7dbddd3e-4eed-bde7-4b58-8bfa6cf7c96e@mikesdriveway.com>

I've installed freeswitch on a Raspberry PI 3 with pretty much bog stock 
config. The only change I've done is to delete the ipv6 profiles and 
changed the default password. I've connected a few different clients, 3 
soft phones and 2 ATAs. When making calls with some of the clients the 
calls are dropped after 32 seconds. For other clients the calls will 
proceed for half an hour no issues. I have absolutely no complexity 
here, it's all internal phones on the local /24 lan. There are no 
firewalls, traffic only goes via a switch. I have no external SIP 
gateway, none of the clients are behind a NAT. I've done extensive 
googling and everything I find talks about NAT issues and VPNs, multiple 
subnets etc. I'm not running any of that. Packet capture and more 
details below.


Some of the things I've tried

- complete OS and freeswitch reinstall on the PI

- disable IPv6 on the PI (made a big improvement, now the remote phone 
rings instantly instead of long delay)

- full OS update before installing freeswitch

- tried clients with both UDP and TCP

- install freeswitch on CentOS 7.

- Disable SIP ALG on router (traffic not going through router anyway)

- Tried all devices on a different network switch

- Tried client options like "use rport" etc




What works:

- Cisco SPA3102 ATA

- Voiper latest version on Windows


What doesn't work

- Latest Voiper on android

- Older version of voiper on windows

- Billion 7404VGP

- Android version of Voip by antisip



TShark capture of a call dropping. IPs are:

192.168.1.245 Freeswitch

192.168.1.225 Windows PC with older copy of voiper

192.168.1.202 Cisco ATA


     1 0.000000000 192.168.1.225 â 192.168.1.245 SIP/SDP 1038 Request: 
INVITE sip:1004 at 192.168.1.245;transport=UDP |
     2 0.005573031 192.168.1.245 â 192.168.1.225 SIP 917 Status: 407 
Proxy Authentication Required |
     3 0.011167520 192.168.1.225 â 192.168.1.245 SIP 408 Request: ACK 
sip:1004 at 192.168.1.245;transport=UDP |
     4 0.016217586 192.168.1.225 â 192.168.1.245 SIP/SDP 1314 Request: 
INVITE sip:1004 at 192.168.1.245;transport=UDP |
     5 0.037157271 192.168.1.245 â 192.168.1.225 SIP 413 Status: 100 
Trying |
     6 0.094830064 192.168.1.245 â 192.168.1.202 SIP/SDP 1283 Request: 
INVITE sip:1004 at 192.168.1.202:5083;transport=tcp |
     7 0.115792718 192.168.1.202 â 192.168.1.245 SIP 365 Status: 100 
Trying |
     8 0.115932040 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083 
[ACK] Seq=1230 Ack=312 Win=63784 Len=0
     9 0.124784941 192.168.1.202 â 192.168.1.245 SIP 526 Status: 180 
Ringing |
    10 0.124863118 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083 
[ACK] Seq=1230 Ack=784 Win=63784 Len=0
    11 0.151217241 192.168.1.245 â 192.168.1.225 SIP/SDP 1205 Status: 
183 Session Progress |
    12 2.263254019 192.168.1.202 â 192.168.1.245 SIP/SDP 901 Status: 200 
OK |
    13 2.263406934 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083 
[ACK] Seq=1230 Ack=1631 Win=63784 Len=0
    14 2.272446033 192.168.1.245 â 192.168.1.202 SIP 471 Request: ACK 
sip:1004 at 192.168.1.202:5083;transport=tcp |
    15 2.284211777 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 
200 OK |
    16 2.469876726 192.168.1.202 â 192.168.1.245 TCP 60 5083 â 5060 
[ACK] Seq=1631 Ack=1647 Win=16000 Len=0
    17 2.784653094 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 
200 OK |
    18 3.785916037 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 
200 OK |
    19 5.787085997 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 
200 OK |
    20 8.903231371 192.168.1.245 â 192.168.1.225 TCP 54 5060 â 56460 
[ACK] Seq=1 Ack=1 Win=501 Len=0
    21 8.903781836 192.168.1.225 â 192.168.1.245 TCP 60 [TCP ACKed 
unseen segment] 56460 â 5060 [ACK] Seq=1 Ack=2 Win=8212 Len=0
    22 9.788128841 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 
200 OK |
    23 13.788284085 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 
200 OK |
    24 17.788442453 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 
200 OK |
    25 20.361180947 192.168.1.225 â 192.168.1.245 UDP 60 5060 â 5060 Len=4
    26 21.788537538 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 
200 OK |
    27 25.788690954 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 
200 OK |
    28 29.789847278 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 
200 OK |
    29 32.583226442 192.168.1.245 â 192.168.1.202 TCP 54 [TCP 
Keep-Alive] 5060 â 5083 [ACK] Seq=1646 Ack=1631 Win=63784 Len=0
    30 32.584076227 192.168.1.202 â 192.168.1.245 TCP 60 [TCP Keep-Alive 
ACK] 5083 â 5060 [ACK] Seq=1631 Ack=1647 Win=16000 Len=0
    31 33.790019961 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 
200 OK |
    32 34.284637050 192.168.1.245 â 192.168.1.225 SIP 704 Request: BYE 
sip:1002 at 192.168.1.225:5060;transport=UDP |
    33 34.327728390 192.168.1.245 â 192.168.1.202 SIP 672 Request: BYE 
sip:1004 at 192.168.1.202:5083;transport=tcp |
    34 34.344437951 192.168.1.202 â 192.168.1.245 SIP 381 Status: 200 OK |
    35 34.393228415 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083 
[ACK] Seq=2265 Ack=1958 Win=63784 Len=0
    36 34.422931003 192.168.1.225 â 192.168.1.245 SIP 466 Status: 200 OK |
    37 38.352414375 192.168.1.225 â 192.168.1.245 TCP 60 [TCP ACKed 
unseen segment] 56460 â 5060 [PSH, ACK] Seq=1 Ack=2 Win=8212 Len=4
    38 38.352610259 192.168.1.245 â 192.168.1.225 TCP 56 [TCP Previous 
segment not captured] 5060 â 56460 [PSH, ACK] Seq=2 Ack=5 Win=501 Len=2
    39 38.393526512 192.168.1.225 â 192.168.1.245 TCP 60 [TCP ACKed 
unseen segment] 56460 â 5060 [ACK] Seq=5 Ack=4 Win=8212 Len=0

-- 
This email has been checked for viruses by AVG.
https://www.avg.com



From kumar94905 at gmail.com  Mon Apr  6 10:05:59 2020
From: kumar94905 at gmail.com (kumar uppu)
Date: Mon, 6 Apr 2020 15:35:59 +0530
Subject: [Freeswitch-users] Call not connecting
Message-ID: <CA+y_vMa5PZV+JJEwj-NCAKq+wBejoSq1jctENJQ-uHcSjLBbzg@mail.gmail.com>

HI All,

Freeswitch sending INVITE with following details

INVITE sip:709 at 49.204.176.126:40067;ob SIP/2.0

Via: SIP/2.0/UDP 52.205.194.64:5080;rport;branch=z9hG4bK0KeS91cvXUjtm


But some how while sending INVITE Registration contact is updating with

Contact: <sip:709 at 49.204.176.126:40241;ob>

Can any please tell me what i am missing?


Thanks,
Kumar


-- 
Kumar
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200406/175a65f9/attachment.html>

From daveh at drachtio.org  Tue Apr  7 02:13:35 2020
From: daveh at drachtio.org (David Horton)
Date: Mon, 6 Apr 2020 22:13:35 -0400
Subject: [Freeswitch-users] Freeswitch drops calls after 32 seconds
In-Reply-To: <7dbddd3e-4eed-bde7-4b58-8bfa6cf7c96e@mikesdriveway.com>
References: <mailman.2146.1585916092.1226.freeswitch-users@lists.freeswitch.org>
 <7dbddd3e-4eed-bde7-4b58-8bfa6cf7c96e@mikesdriveway.com>
Message-ID: <977169DB-C399-4C8C-9C01-1E38EA0EC237@drachtio.org>

Call drops after 32s are almost always caused by an ACK timeout.  Given the retransmits shown in your listing where freeswitch is resending the 200 OK and the client is resending the ACK, this does appear to be the problem.  Either the ACK is not reaching freeswitch or (more likely) it is not being matched to the the 200 OK response.  Hard to say more without seeing the actual messages.

Dave

On Apr 3, 2020, at 9:11 AM, MikeKulls <freeswitch at mikesdriveway.com> wrote:

I've installed freeswitch on a Raspberry PI 3 with pretty much bog stock config. The only change I've done is to delete the ipv6 profiles and changed the default password. I've connected a few different clients, 3 soft phones and 2 ATAs. When making calls with some of the clients the calls are dropped after 32 seconds. For other clients the calls will proceed for half an hour no issues. I have absolutely no complexity here, it's all internal phones on the local /24 lan. There are no firewalls, traffic only goes via a switch. I have no external SIP gateway, none of the clients are behind a NAT. I've done extensive googling and everything I find talks about NAT issues and VPNs, multiple subnets etc. I'm not running any of that. Packet capture and more details below.


Some of the things I've tried

- complete OS and freeswitch reinstall on the PI

- disable IPv6 on the PI (made a big improvement, now the remote phone rings instantly instead of long delay)

- full OS update before installing freeswitch

- tried clients with both UDP and TCP

- install freeswitch on CentOS 7.

- Disable SIP ALG on router (traffic not going through router anyway)

- Tried all devices on a different network switch

- Tried client options like "use rport" etc




What works:

- Cisco SPA3102 ATA

- Voiper latest version on Windows


What doesn't work

- Latest Voiper on android

- Older version of voiper on windows

- Billion 7404VGP

- Android version of Voip by antisip



TShark capture of a call dropping. IPs are:

192.168.1.245 Freeswitch

192.168.1.225 Windows PC with older copy of voiper

192.168.1.202 Cisco ATA


    1 0.000000000 192.168.1.225 â 192.168.1.245 SIP/SDP 1038 Request: INVITE sip:1004 at 192.168.1.245;transport=UDP |
    2 0.005573031 192.168.1.245 â 192.168.1.225 SIP 917 Status: 407 Proxy Authentication Required |
    3 0.011167520 192.168.1.225 â 192.168.1.245 SIP 408 Request: ACK sip:1004 at 192.168.1.245;transport=UDP |
    4 0.016217586 192.168.1.225 â 192.168.1.245 SIP/SDP 1314 Request: INVITE sip:1004 at 192.168.1.245;transport=UDP |
    5 0.037157271 192.168.1.245 â 192.168.1.225 SIP 413 Status: 100 Trying |
    6 0.094830064 192.168.1.245 â 192.168.1.202 SIP/SDP 1283 Request: INVITE sip:1004 at 192.168.1.202:5083;transport=tcp |
    7 0.115792718 192.168.1.202 â 192.168.1.245 SIP 365 Status: 100 Trying |
    8 0.115932040 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083 [ACK] Seq=1230 Ack=312 Win=63784 Len=0
    9 0.124784941 192.168.1.202 â 192.168.1.245 SIP 526 Status: 180 Ringing |
   10 0.124863118 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083 [ACK] Seq=1230 Ack=784 Win=63784 Len=0
   11 0.151217241 192.168.1.245 â 192.168.1.225 SIP/SDP 1205 Status: 183 Session Progress |
   12 2.263254019 192.168.1.202 â 192.168.1.245 SIP/SDP 901 Status: 200 OK |
   13 2.263406934 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083 [ACK] Seq=1230 Ack=1631 Win=63784 Len=0
   14 2.272446033 192.168.1.245 â 192.168.1.202 SIP 471 Request: ACK sip:1004 at 192.168.1.202:5083;transport=tcp |
   15 2.284211777 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK |
   16 2.469876726 192.168.1.202 â 192.168.1.245 TCP 60 5083 â 5060 [ACK] Seq=1631 Ack=1647 Win=16000 Len=0
   17 2.784653094 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK |
   18 3.785916037 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK |
   19 5.787085997 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK |
   20 8.903231371 192.168.1.245 â 192.168.1.225 TCP 54 5060 â 56460 [ACK] Seq=1 Ack=1 Win=501 Len=0
   21 8.903781836 192.168.1.225 â 192.168.1.245 TCP 60 [TCP ACKed unseen segment] 56460 â 5060 [ACK] Seq=1 Ack=2 Win=8212 Len=0
   22 9.788128841 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK |
   23 13.788284085 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK |
   24 17.788442453 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK |
   25 20.361180947 192.168.1.225 â 192.168.1.245 UDP 60 5060 â 5060 Len=4
   26 21.788537538 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK |
   27 25.788690954 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK |
   28 29.789847278 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK |
   29 32.583226442 192.168.1.245 â 192.168.1.202 TCP 54 [TCP Keep-Alive] 5060 â 5083 [ACK] Seq=1646 Ack=1631 Win=63784 Len=0
   30 32.584076227 192.168.1.202 â 192.168.1.245 TCP 60 [TCP Keep-Alive ACK] 5083 â 5060 [ACK] Seq=1631 Ack=1647 Win=16000 Len=0
   31 33.790019961 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK |
   32 34.284637050 192.168.1.245 â 192.168.1.225 SIP 704 Request: BYE sip:1002 at 192.168.1.225:5060;transport=UDP |
   33 34.327728390 192.168.1.245 â 192.168.1.202 SIP 672 Request: BYE sip:1004 at 192.168.1.202:5083;transport=tcp |
   34 34.344437951 192.168.1.202 â 192.168.1.245 SIP 381 Status: 200 OK |
   35 34.393228415 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083 [ACK] Seq=2265 Ack=1958 Win=63784 Len=0
   36 34.422931003 192.168.1.225 â 192.168.1.245 SIP 466 Status: 200 OK |
   37 38.352414375 192.168.1.225 â 192.168.1.245 TCP 60 [TCP ACKed unseen segment] 56460 â 5060 [PSH, ACK] Seq=1 Ack=2 Win=8212 Len=4
   38 38.352610259 192.168.1.245 â 192.168.1.225 TCP 56 [TCP Previous segment not captured] 5060 â 56460 [PSH, ACK] Seq=2 Ack=5 Win=501 Len=2
   39 38.393526512 192.168.1.225 â 192.168.1.245 TCP 60 [TCP ACKed unseen segment] 56460 â 5060 [ACK] Seq=5 Ack=4 Win=8212 Len=0

-- 
This email has been checked for viruses by AVG.
https://www.avg.com


_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

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

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

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



From krice at freeswitch.org  Tue Apr  7 02:28:42 2020
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 06 Apr 2020 21:28:42 -0500
Subject: [Freeswitch-users] Freeswitch drops calls after 32 seconds
In-Reply-To: <7dbddd3e-4eed-bde7-4b58-8bfa6cf7c96e@mikesdriveway.com>
References: <mailman.2146.1585916092.1226.freeswitch-users@lists.freeswitch.org>
 <7dbddd3e-4eed-bde7-4b58-8bfa6cf7c96e@mikesdriveway.com>
Message-ID: <3498792D-08F1-41DF-A4C9-2396B9EFC5F2@freeswitch.org>

Firewall? For whatever reason FS is not getting the ACKs from the 192.168.1.225 machine to the 200OK telling it the call was answered. You can see this where the FS box sends multiple 200 Oks.  When FS doesn’t get the ACK within 32 seconds it kills the call.

On 4/6/20, 8:39 PM, "FreeSWITCH-users on behalf of MikeKulls" <freeswitch-users-bounces at lists.freeswitch.org on behalf of freeswitch at mikesdriveway.com> wrote:

    I've installed freeswitch on a Raspberry PI 3 with pretty much bog stock 
    config. The only change I've done is to delete the ipv6 profiles and 
    changed the default password. I've connected a few different clients, 3 
    soft phones and 2 ATAs. When making calls with some of the clients the 
    calls are dropped after 32 seconds. For other clients the calls will 
    proceed for half an hour no issues. I have absolutely no complexity 
    here, it's all internal phones on the local /24 lan. There are no 
    firewalls, traffic only goes via a switch. I have no external SIP 
    gateway, none of the clients are behind a NAT. I've done extensive 
    googling and everything I find talks about NAT issues and VPNs, multiple 
    subnets etc. I'm not running any of that. Packet capture and more 
    details below.
    
    
    Some of the things I've tried
    
    - complete OS and freeswitch reinstall on the PI
    
    - disable IPv6 on the PI (made a big improvement, now the remote phone 
    rings instantly instead of long delay)
    
    - full OS update before installing freeswitch
    
    - tried clients with both UDP and TCP
    
    - install freeswitch on CentOS 7.
    
    - Disable SIP ALG on router (traffic not going through router anyway)
    
    - Tried all devices on a different network switch
    
    - Tried client options like "use rport" etc
    
    
    
    
    What works:
    
    - Cisco SPA3102 ATA
    
    - Voiper latest version on Windows
    
    
    What doesn't work
    
    - Latest Voiper on android
    
    - Older version of voiper on windows
    
    - Billion 7404VGP
    
    - Android version of Voip by antisip
    
    
    
    TShark capture of a call dropping. IPs are:
    
    192.168.1.245 Freeswitch
    
    192.168.1.225 Windows PC with older copy of voiper
    
    192.168.1.202 Cisco ATA
    
    
         1 0.000000000 192.168.1.225 â 192.168.1.245 SIP/SDP 1038 Request: 
    INVITE sip:1004 at 192.168.1.245;transport=UDP |
         2 0.005573031 192.168.1.245 â 192.168.1.225 SIP 917 Status: 407 
    Proxy Authentication Required |
         3 0.011167520 192.168.1.225 â 192.168.1.245 SIP 408 Request: ACK 
    sip:1004 at 192.168.1.245;transport=UDP |
         4 0.016217586 192.168.1.225 â 192.168.1.245 SIP/SDP 1314 Request: 
    INVITE sip:1004 at 192.168.1.245;transport=UDP |
         5 0.037157271 192.168.1.245 â 192.168.1.225 SIP 413 Status: 100 
    Trying |
         6 0.094830064 192.168.1.245 â 192.168.1.202 SIP/SDP 1283 Request: 
    INVITE sip:1004 at 192.168.1.202:5083;transport=tcp |
         7 0.115792718 192.168.1.202 â 192.168.1.245 SIP 365 Status: 100 
    Trying |
         8 0.115932040 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083 
    [ACK] Seq=1230 Ack=312 Win=63784 Len=0
         9 0.124784941 192.168.1.202 â 192.168.1.245 SIP 526 Status: 180 
    Ringing |
        10 0.124863118 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083 
    [ACK] Seq=1230 Ack=784 Win=63784 Len=0
        11 0.151217241 192.168.1.245 â 192.168.1.225 SIP/SDP 1205 Status: 
    183 Session Progress |
        12 2.263254019 192.168.1.202 â 192.168.1.245 SIP/SDP 901 Status: 200 
    OK |
        13 2.263406934 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083 
    [ACK] Seq=1230 Ack=1631 Win=63784 Len=0
        14 2.272446033 192.168.1.245 â 192.168.1.202 SIP 471 Request: ACK 
    sip:1004 at 192.168.1.202:5083;transport=tcp |
        15 2.284211777 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 
    200 OK |
        16 2.469876726 192.168.1.202 â 192.168.1.245 TCP 60 5083 â 5060 
    [ACK] Seq=1631 Ack=1647 Win=16000 Len=0
        17 2.784653094 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 
    200 OK |
        18 3.785916037 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 
    200 OK |
        19 5.787085997 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 
    200 OK |
        20 8.903231371 192.168.1.245 â 192.168.1.225 TCP 54 5060 â 56460 
    [ACK] Seq=1 Ack=1 Win=501 Len=0
        21 8.903781836 192.168.1.225 â 192.168.1.245 TCP 60 [TCP ACKed 
    unseen segment] 56460 â 5060 [ACK] Seq=1 Ack=2 Win=8212 Len=0
        22 9.788128841 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 
    200 OK |
        23 13.788284085 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 
    200 OK |
        24 17.788442453 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 
    200 OK |
        25 20.361180947 192.168.1.225 â 192.168.1.245 UDP 60 5060 â 5060 Len=4
        26 21.788537538 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 
    200 OK |
        27 25.788690954 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 
    200 OK |
        28 29.789847278 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 
    200 OK |
        29 32.583226442 192.168.1.245 â 192.168.1.202 TCP 54 [TCP 
    Keep-Alive] 5060 â 5083 [ACK] Seq=1646 Ack=1631 Win=63784 Len=0
        30 32.584076227 192.168.1.202 â 192.168.1.245 TCP 60 [TCP Keep-Alive 
    ACK] 5083 â 5060 [ACK] Seq=1631 Ack=1647 Win=16000 Len=0
        31 33.790019961 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 
    200 OK |
        32 34.284637050 192.168.1.245 â 192.168.1.225 SIP 704 Request: BYE 
    sip:1002 at 192.168.1.225:5060;transport=UDP |
        33 34.327728390 192.168.1.245 â 192.168.1.202 SIP 672 Request: BYE 
    sip:1004 at 192.168.1.202:5083;transport=tcp |
        34 34.344437951 192.168.1.202 â 192.168.1.245 SIP 381 Status: 200 OK |
        35 34.393228415 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083 
    [ACK] Seq=2265 Ack=1958 Win=63784 Len=0
        36 34.422931003 192.168.1.225 â 192.168.1.245 SIP 466 Status: 200 OK |
        37 38.352414375 192.168.1.225 â 192.168.1.245 TCP 60 [TCP ACKed 
    unseen segment] 56460 â 5060 [PSH, ACK] Seq=1 Ack=2 Win=8212 Len=4
        38 38.352610259 192.168.1.245 â 192.168.1.225 TCP 56 [TCP Previous 
    segment not captured] 5060 â 56460 [PSH, ACK] Seq=2 Ack=5 Win=501 Len=2
        39 38.393526512 192.168.1.225 â 192.168.1.245 TCP 60 [TCP ACKed 
    unseen segment] 56460 â 5060 [ACK] Seq=5 Ack=4 Win=8212 Len=0
    
    -- 
    This email has been checked for viruses by AVG.
    https://www.avg.com
    
    
    _________________________________________________________________________
    
    The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
    Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
    Build your next product on our scalable cloud platform.
    
    Join our online community to chat in real time https://signalwire.community
    
    Professional FreeSWITCH Services
    sales at freeswitch.com
    https://freeswitch.com
    
    Official FreeSWITCH Sites
    https://freeswitch.com/oss
    https://freeswitch.org/confluence
    https://cluecon.com
    
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    https://freeswitch.com




From loidang at hoiio.com  Tue Apr  7 02:35:15 2020
From: loidang at hoiio.com (Loi Dang)
Date: Tue, 7 Apr 2020 09:35:15 +0700
Subject: [Freeswitch-users] SIP 302 Redirect
In-Reply-To: <CALv+5v5+01U_ecGdFdpEN1r+T3L5WigzmxC6G+ZZtZd9q9zhAw@mail.gmail.com>
References: <CALv+5v7oKEdo9W1iw3icMgbasZ=nNzbXE4TMVgrm5yOYu3X8UA@mail.gmail.com>
 <CADMnEC9ada3K-j1pUa9gCShwq7sX8mKf20KPq6R93fiVDb5=Lg@mail.gmail.com>
 <CALv+5v6D7Mk9g3M9XOeY=HhwTpzuDNuL-dvwerYNiZf==RfdKw@mail.gmail.com>
 <CADMnEC8YuZc103wKB8JtHNVNW_AmkEnhaiyiP8PteHHPkppteQ@mail.gmail.com>
 <CALv+5v5+01U_ecGdFdpEN1r+T3L5WigzmxC6G+ZZtZd9q9zhAw@mail.gmail.com>
Message-ID: <CADMnEC8p9Y9LQbvgEK_1r9zK2nerAzm8x6chiKyMVsbRG4+zGg@mail.gmail.com>

You may want to deal with FS channel variable here, especially with
export_vars variable and set & export functions
https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables

Also take a look at sip_h_*, sip_rh_*, and sip_ph headers for dealing with
SIP header (request & response)

rgds,
Loi Dang

On Mon, Apr 6, 2020 at 6:05 PM Giri <girish.dharmaraj at gmail.com> wrote:

> Hello Sir,
>
> Noted with thanks. Relaly appreciate your response.  I am trying it but I
> need another clarification if you can see the below message received from
> the outbound on the 302 SIP , I have an header Subject: Callinfo, is it
> possible to store this Callinfo to a variable and pass it to the inbound
> side as it is without making any changes to the Subject header .
>
> SIP/2.0 302 Moved Temporarily
>    Via: SIP/2.0/UDP 192.168.200.18:5080
> ;rport=5080;branch=z9hG4bK2Z7c6ZaXSQt5a
>    Contact: <sip:11111 at XX.XX.XX.XX>
>    To: <sip:1234567895 at 192.168.200.44:30007>;tag=aa14db7c
>    From: "104"<sip:FreeSWITCH at 192.168.200.44:30007>;tag=gHg6SeSDN569j
>    Call-ID: a80638c1-f296-1238-21ab-3417ebe52801
>    CSeq: 18531190 INVITE
>    Subject:
> callinfo=012020001A01316P100000000000448048898116681921495000035040100100100586400035010100100100586400035020100100100586400031020100100100586400031020100100100586400010
>    User-Agent: hmd/2.0
>    Content-Length: 0
>
> With Best Regards,
> Girish Dharmaraj
>
>
> On Mon, Apr 6, 2020 at 10:51 AM Loi Dang <loidang at hoiio.com> wrote:
>
>> try * wildcard matching in your `redirected` context's extension, then do
>> a <action application="redirect" data="sip:${destination_number}@$${sip_ip}"/>
>> in the end.
>> Hope this helps.
>>
>> rgds,
>> Loi Dang
>>
>> On Sun, Apr 5, 2020 at 6:26 PM Giri <girish.dharmaraj at gmail.com> wrote:
>>
>>> Hello Sir,
>>>
>>> I tried that, but it just goes into the redirect context and fails
>>> finding no match and call is released with 503.
>>>
>>> On Sat, Apr 4, 2020, 23:36 Loi Dang <loidang at hoiio.com> wrote:
>>>
>>>> Hi, try manual-redirect option in SIP profile?
>>>> rgds,
>>>> Loi Dang
>>>>
>>>> On Sat, Apr 4, 2020, 10:12 PM Giri <girish.dharmaraj at gmail.com> wrote:
>>>>
>>>>> Good Day, I have a scenario where the Outbound GW sends SIP 302 and
>>>>> all I need is to send the SIP 302 message to the Inbound.  No Routing needs
>>>>> to take place, is this possible?
>>>>>
>>>>> With Best Regards,
>>>>> Girish Dharmaraj
>>>>>
>>>>> _________________________________________________________________________
>>>>>
>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>> https://signalwire.com
>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>> services.
>>>>> Build your next product on our scalable cloud platform.
>>>>>
>>>>> Join our online community to chat in real time
>>>>> https://signalwire.community
>>>>>
>>>>> Professional FreeSWITCH Services
>>>>> sales at freeswitch.com
>>>>> https://freeswitch.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> https://freeswitch.com/oss
>>>>> https://freeswitch.org/confluence
>>>>> https://cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> https://freeswitch.com
>>>>
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200407/010b79d7/attachment-0001.html>

From girish.dharmaraj at gmail.com  Tue Apr  7 02:51:17 2020
From: girish.dharmaraj at gmail.com (Giri)
Date: Tue, 7 Apr 2020 10:51:17 +0800
Subject: [Freeswitch-users] SIP 302 Redirect
In-Reply-To: <CADMnEC8p9Y9LQbvgEK_1r9zK2nerAzm8x6chiKyMVsbRG4+zGg@mail.gmail.com>
References: <CALv+5v7oKEdo9W1iw3icMgbasZ=nNzbXE4TMVgrm5yOYu3X8UA@mail.gmail.com>
 <CADMnEC9ada3K-j1pUa9gCShwq7sX8mKf20KPq6R93fiVDb5=Lg@mail.gmail.com>
 <CALv+5v6D7Mk9g3M9XOeY=HhwTpzuDNuL-dvwerYNiZf==RfdKw@mail.gmail.com>
 <CADMnEC8YuZc103wKB8JtHNVNW_AmkEnhaiyiP8PteHHPkppteQ@mail.gmail.com>
 <CALv+5v5+01U_ecGdFdpEN1r+T3L5WigzmxC6G+ZZtZd9q9zhAw@mail.gmail.com>
 <CADMnEC8p9Y9LQbvgEK_1r9zK2nerAzm8x6chiKyMVsbRG4+zGg@mail.gmail.com>
Message-ID: <CALv+5v6F1RHa4hTF=mPEmzr7Y4wqaOOzafJmkKJzwedgLpRMfQ@mail.gmail.com>

Hello Sir,

Thanks a lot again, I did try that but looks like if the *SIP customer *
header in the response 302 is not prefixed with X- or P-
FS is unable to parse it and store it to a channel variable. I am not sure
if my understanding is right, but this is what I found out. Is that correct?

With Best Regards,
Girish Dharmaraj


On Tue, Apr 7, 2020 at 10:36 AM Loi Dang <loidang at hoiio.com> wrote:

> You may want to deal with FS channel variable here, especially with
> export_vars variable and set & export functions
> https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables
>
> Also take a look at sip_h_*, sip_rh_*, and sip_ph headers for dealing
> with SIP header (request & response)
>
> rgds,
> Loi Dang
>
> On Mon, Apr 6, 2020 at 6:05 PM Giri <girish.dharmaraj at gmail.com> wrote:
>
>> Hello Sir,
>>
>> Noted with thanks. Relaly appreciate your response.  I am trying it but I
>> need another clarification if you can see the below message received from
>> the outbound on the 302 SIP , I have an header Subject: Callinfo, is it
>> possible to store this Callinfo to a variable and pass it to the inbound
>> side as it is without making any changes to the Subject header .
>>
>> SIP/2.0 302 Moved Temporarily
>>    Via: SIP/2.0/UDP 192.168.200.18:5080
>> ;rport=5080;branch=z9hG4bK2Z7c6ZaXSQt5a
>>    Contact: <sip:11111 at XX.XX.XX.XX>
>>    To: <sip:1234567895 at 192.168.200.44:30007>;tag=aa14db7c
>>    From: "104"<sip:FreeSWITCH at 192.168.200.44:30007>;tag=gHg6SeSDN569j
>>    Call-ID: a80638c1-f296-1238-21ab-3417ebe52801
>>    CSeq: 18531190 INVITE
>>    Subject:
>> callinfo=012020001A01316P100000000000448048898116681921495000035040100100100586400035010100100100586400035020100100100586400031020100100100586400031020100100100586400010
>>    User-Agent: hmd/2.0
>>    Content-Length: 0
>>
>> With Best Regards,
>> Girish Dharmaraj
>>
>>
>> On Mon, Apr 6, 2020 at 10:51 AM Loi Dang <loidang at hoiio.com> wrote:
>>
>>> try * wildcard matching in your `redirected` context's extension, then
>>> do a <action application="redirect" data="sip:${destination_number}@$${sip_ip}"/>
>>> in the end.
>>> Hope this helps.
>>>
>>> rgds,
>>> Loi Dang
>>>
>>> On Sun, Apr 5, 2020 at 6:26 PM Giri <girish.dharmaraj at gmail.com> wrote:
>>>
>>>> Hello Sir,
>>>>
>>>> I tried that, but it just goes into the redirect context and fails
>>>> finding no match and call is released with 503.
>>>>
>>>> On Sat, Apr 4, 2020, 23:36 Loi Dang <loidang at hoiio.com> wrote:
>>>>
>>>>> Hi, try manual-redirect option in SIP profile?
>>>>> rgds,
>>>>> Loi Dang
>>>>>
>>>>> On Sat, Apr 4, 2020, 10:12 PM Giri <girish.dharmaraj at gmail.com> wrote:
>>>>>
>>>>>> Good Day, I have a scenario where the Outbound GW sends SIP 302 and
>>>>>> all I need is to send the SIP 302 message to the Inbound.  No Routing needs
>>>>>> to take place, is this possible?
>>>>>>
>>>>>> With Best Regards,
>>>>>> Girish Dharmaraj
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>>
>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>> https://signalwire.com
>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>>> services.
>>>>>> Build your next product on our scalable cloud platform.
>>>>>>
>>>>>> Join our online community to chat in real time
>>>>>> https://signalwire.community
>>>>>>
>>>>>> Professional FreeSWITCH Services
>>>>>> sales at freeswitch.com
>>>>>> https://freeswitch.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> https://freeswitch.com/oss
>>>>>> https://freeswitch.org/confluence
>>>>>> https://cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> https://freeswitch.com
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>>
>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>> https://signalwire.com
>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>> services.
>>>>> Build your next product on our scalable cloud platform.
>>>>>
>>>>> Join our online community to chat in real time
>>>>> https://signalwire.community
>>>>>
>>>>> Professional FreeSWITCH Services
>>>>> sales at freeswitch.com
>>>>> https://freeswitch.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> https://freeswitch.com/oss
>>>>> https://freeswitch.org/confluence
>>>>> https://cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> https://freeswitch.com
>>>>
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200407/d2739522/attachment-0001.html>

From david.villasmil.work at gmail.com  Tue Apr  7 08:48:58 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 7 Apr 2020 09:48:58 +0100
Subject: [Freeswitch-users] Freeswitch drops calls after 32 seconds
In-Reply-To: <3498792D-08F1-41DF-A4C9-2396B9EFC5F2@freeswitch.org>
References: <mailman.2146.1585916092.1226.freeswitch-users@lists.freeswitch.org>
 <7dbddd3e-4eed-bde7-4b58-8bfa6cf7c96e@mikesdriveway.com>
 <3498792D-08F1-41DF-A4C9-2396B9EFC5F2@freeswitch.org>
Message-ID: <CAFGRPVrOzVuVAuOJ55UGqa_s8PDUs_7VmhoUcj3wK1gducfaDA@mail.gmail.com>

+1 plus check the phone are not using STUN and are configured to use their
own IPs, I’ve seen this a lot.

On Tue, 7 Apr 2020 at 04:03, Ken Rice <krice at freeswitch.org> wrote:

> Firewall? For whatever reason FS is not getting the ACKs from the
> 192.168.1.225 machine to the 200OK telling it the call was answered. You
> can see this where the FS box sends multiple 200 Oks.  When FS doesn’t get
> the ACK within 32 seconds it kills the call.
>
> On 4/6/20, 8:39 PM, "FreeSWITCH-users on behalf of MikeKulls" <
> freeswitch-users-bounces at lists.freeswitch.org on behalf of
> freeswitch at mikesdriveway.com> wrote:
>
>     I've installed freeswitch on a Raspberry PI 3 with pretty much bog
> stock
>     config. The only change I've done is to delete the ipv6 profiles and
>     changed the default password. I've connected a few different clients,
> 3
>     soft phones and 2 ATAs. When making calls with some of the clients the
>     calls are dropped after 32 seconds. For other clients the calls will
>     proceed for half an hour no issues. I have absolutely no complexity
>     here, it's all internal phones on the local /24 lan. There are no
>     firewalls, traffic only goes via a switch. I have no external SIP
>     gateway, none of the clients are behind a NAT. I've done extensive
>     googling and everything I find talks about NAT issues and VPNs,
> multiple
>     subnets etc. I'm not running any of that. Packet capture and more
>     details below.
>
>
>     Some of the things I've tried
>
>     - complete OS and freeswitch reinstall on the PI
>
>     - disable IPv6 on the PI (made a big improvement, now the remote phone
>     rings instantly instead of long delay)
>
>     - full OS update before installing freeswitch
>
>     - tried clients with both UDP and TCP
>
>     - install freeswitch on CentOS 7.
>
>     - Disable SIP ALG on router (traffic not going through router anyway)
>
>     - Tried all devices on a different network switch
>
>     - Tried client options like "use rport" etc
>
>
>
>
>     What works:
>
>     - Cisco SPA3102 ATA
>
>     - Voiper latest version on Windows
>
>
>     What doesn't work
>
>     - Latest Voiper on android
>
>     - Older version of voiper on windows
>
>     - Billion 7404VGP
>
>     - Android version of Voip by antisip
>
>
>
>     TShark capture of a call dropping. IPs are:
>
>     192.168.1.245 Freeswitch
>
>     192.168.1.225 Windows PC with older copy of voiper
>
>     192.168.1.202 Cisco ATA
>
>
>          1 0.000000000 192.168.1.225 â 192.168.1.245 SIP/SDP 1038 Request:
>     INVITE sip:1004 at 192.168.1.245;transport=UDP |
>          2 0.005573031 192.168.1.245 â 192.168.1.225 SIP 917 Status: 407
>     Proxy Authentication Required |
>          3 0.011167520 192.168.1.225 â 192.168.1.245 SIP 408 Request: ACK
>     sip:1004 at 192.168.1.245;transport=UDP |
>          4 0.016217586 192.168.1.225 â 192.168.1.245 SIP/SDP 1314 Request:
>     INVITE sip:1004 at 192.168.1.245;transport=UDP |
>          5 0.037157271 192.168.1.245 â 192.168.1.225 SIP 413 Status: 100
>     Trying |
>          6 0.094830064 192.168.1.245 â 192.168.1.202 SIP/SDP 1283 Request:
>     INVITE sip:1004 at 192.168.1.202:5083;transport=tcp |
>          7 0.115792718 192.168.1.202 â 192.168.1.245 SIP 365 Status: 100
>     Trying |
>          8 0.115932040 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083
>     [ACK] Seq=1230 Ack=312 Win=63784 Len=0
>          9 0.124784941 192.168.1.202 â 192.168.1.245 SIP 526 Status: 180
>     Ringing |
>         10 0.124863118 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083
>     [ACK] Seq=1230 Ack=784 Win=63784 Len=0
>         11 0.151217241 192.168.1.245 â 192.168.1.225 SIP/SDP 1205 Status:
>     183 Session Progress |
>         12 2.263254019 192.168.1.202 â 192.168.1.245 SIP/SDP 901 Status:
> 200
>     OK |
>         13 2.263406934 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083
>     [ACK] Seq=1230 Ack=1631 Win=63784 Len=0
>         14 2.272446033 192.168.1.245 â 192.168.1.202 SIP 471 Request: ACK
>     sip:1004 at 192.168.1.202:5083;transport=tcp |
>         15 2.284211777 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status:
>     200 OK |
>         16 2.469876726 192.168.1.202 â 192.168.1.245 TCP 60 5083 â 5060
>     [ACK] Seq=1631 Ack=1647 Win=16000 Len=0
>         17 2.784653094 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status:
>     200 OK |
>         18 3.785916037 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status:
>     200 OK |
>         19 5.787085997 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status:
>     200 OK |
>         20 8.903231371 192.168.1.245 â 192.168.1.225 TCP 54 5060 â 56460
>     [ACK] Seq=1 Ack=1 Win=501 Len=0
>         21 8.903781836 192.168.1.225 â 192.168.1.245 TCP 60 [TCP ACKed
>     unseen segment] 56460 â 5060 [ACK] Seq=1 Ack=2 Win=8212 Len=0
>         22 9.788128841 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status:
>     200 OK |
>         23 13.788284085 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status:
>     200 OK |
>         24 17.788442453 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status:
>     200 OK |
>         25 20.361180947 192.168.1.225 â 192.168.1.245 UDP 60 5060 â 5060
> Len=4
>         26 21.788537538 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status:
>     200 OK |
>         27 25.788690954 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status:
>     200 OK |
>         28 29.789847278 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status:
>     200 OK |
>         29 32.583226442 192.168.1.245 â 192.168.1.202 TCP 54 [TCP
>     Keep-Alive] 5060 â 5083 [ACK] Seq=1646 Ack=1631 Win=63784 Len=0
>         30 32.584076227 192.168.1.202 â 192.168.1.245 TCP 60 [TCP
> Keep-Alive
>     ACK] 5083 â 5060 [ACK] Seq=1631 Ack=1647 Win=16000 Len=0
>         31 33.790019961 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status:
>     200 OK |
>         32 34.284637050 192.168.1.245 â 192.168.1.225 SIP 704 Request: BYE
>     sip:1002 at 192.168.1.225:5060;transport=UDP |
>         33 34.327728390 192.168.1.245 â 192.168.1.202 SIP 672 Request: BYE
>     sip:1004 at 192.168.1.202:5083;transport=tcp |
>         34 34.344437951 192.168.1.202 â 192.168.1.245 SIP 381 Status: 200
> OK |
>         35 34.393228415 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083
>     [ACK] Seq=2265 Ack=1958 Win=63784 Len=0
>         36 34.422931003 192.168.1.225 â 192.168.1.245 SIP 466 Status: 200
> OK |
>         37 38.352414375 192.168.1.225 â 192.168.1.245 TCP 60 [TCP ACKed
>     unseen segment] 56460 â 5060 [PSH, ACK] Seq=1 Ack=2 Win=8212 Len=4
>         38 38.352610259 192.168.1.245 â 192.168.1.225 TCP 56 [TCP Previous
>     segment not captured] 5060 â 56460 [PSH, ACK] Seq=2 Ack=5 Win=501 Len=2
>         39 38.393526512 192.168.1.225 â 192.168.1.245 TCP 60 [TCP ACKed
>     unseen segment] 56460 â 5060 [ACK] Seq=5 Ack=4 Win=8212 Len=0
>
>     --
>     This email has been checked for viruses by AVG.
>     https://www.avg.com
>
>
>
> _________________________________________________________________________
>
>     The FreeSWITCH project is sponsored by SignalWire
> https://signalwire.com
>     Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
>     Build your next product on our scalable cloud platform.
>
>     Join our online community to chat in real time
> https://signalwire.community
>
>     Professional FreeSWITCH Services
>     sales at freeswitch.com
>     https://freeswitch.com
>
>     Official FreeSWITCH Sites
>     https://freeswitch.com/oss
>     https://freeswitch.org/confluence
>     https://cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
>     https://freeswitch.com
>
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200407/45dc8171/attachment.html>

From david.villasmil.work at gmail.com  Tue Apr  7 08:50:57 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 7 Apr 2020 09:50:57 +0100
Subject: [Freeswitch-users] Call not connecting
In-Reply-To: <CA+y_vMa5PZV+JJEwj-NCAKq+wBejoSq1jctENJQ-uHcSjLBbzg@mail.gmail.com>
References: <CA+y_vMa5PZV+JJEwj-NCAKq+wBejoSq1jctENJQ-uHcSjLBbzg@mail.gmail.com>
Message-ID: <CAFGRPVpGEvrUcRjDRHeO+F6Uhq680TgDd6_aVa36QSPu62543w@mail.gmail.com>

I have no idea what you are asking.

Can you please elaborate? Like, what version are you using? Is the endpoint
registered? Can you provide signaling traces as well as fs logs?

On Tue, 7 Apr 2020 at 02:54, kumar uppu <kumar94905 at gmail.com> wrote:

> HI All,
>
> Freeswitch sending INVITE with following details
>
> INVITE sip:709 at 49.204.176.126:40067;ob SIP/2.0
>
> Via: SIP/2.0/UDP 52.205.194.64:5080;rport;branch=z9hG4bK0KeS91cvXUjtm
>
>
> But some how while sending INVITE Registration contact is updating with
>
> Contact: <sip:709 at 49.204.176.126:40241;ob>
>
> Can any please tell me what i am missing?
>
>
> Thanks,
> Kumar
>
>
> --
> Kumar
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200407/652efac3/attachment-0001.html>

From thilo at ginkel.com  Wed Apr  8 19:54:31 2020
From: thilo at ginkel.com (Thilo-Alexander Ginkel)
Date: Wed, 8 Apr 2020 21:54:31 +0200
Subject: [Freeswitch-users] execute_on_answer equivalent for A-leg or
	bridged call?
Message-ID: <CANvSZQ-fFJpk2UK4vhq4S6qP1k7fh-eYMeQA5eDkesNVesuCfA@mail.gmail.com>

Hello everyone,

I am attempting to play a sound on both legs after creating a bridge
once the B-leg has answered. execute_on_answer explicitly only works
for the newly answered channel, so no sound is playing on the
(previously already connected) A-leg.

Is there a way to make sure the sound is playing on both legs / the
bridged call?

ATM, I'm doing this (in Lua):

session:execute("bridge", "{origination_callee_id_name='" .. callee_id
.. "'}{execute_on_answer='playback
tone_stream://%(500,0,500)'}user/jitsi@" .. domain_name)

Thanks,
Thilo


From brian at freeswitch.com  Wed Apr  8 21:18:26 2020
From: brian at freeswitch.com (Brian West)
Date: Wed, 8 Apr 2020 16:18:26 -0500
Subject: [Freeswitch-users] execute_on_answer equivalent for A-leg or
 bridged call?
In-Reply-To: <CANvSZQ-fFJpk2UK4vhq4S6qP1k7fh-eYMeQA5eDkesNVesuCfA@mail.gmail.com>
References: <CANvSZQ-fFJpk2UK4vhq4S6qP1k7fh-eYMeQA5eDkesNVesuCfA@mail.gmail.com>
Message-ID: <CAEJMVkDgcHRCHXOsNGC-b42wrQ1SOUK8SCxJ-m05su=vQmw9Rw@mail.gmail.com>

Set the variable on the a-leg
/b


On Wed, Apr 8, 2020 at 3:33 PM Thilo-Alexander Ginkel <thilo at ginkel.com>
wrote:

> Hello everyone,
>
> I am attempting to play a sound on both legs after creating a bridge
> once the B-leg has answered. execute_on_answer explicitly only works
> for the newly answered channel, so no sound is playing on the
> (previously already connected) A-leg.
>
> Is there a way to make sure the sound is playing on both legs / the
> bridged call?
>
> ATM, I'm doing this (in Lua):
>
> session:execute("bridge", "{origination_callee_id_name='" .. callee_id
> .. "'}{execute_on_answer='playback
> tone_stream://%(500,0,500)'}user/jitsi@" .. domain_name)
>
> Thanks,
> Thilo
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 

Brian West | Co-founder and Developer

Need Commercial support? email sales at freeswitch.com

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

Email: brian at freeswitch.com

Mobile: 918-424-9378

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

[image: https://www.facebook.com/signalwireinc?src=email]
<https://www.facebook.com/freeswitch> [image:
https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200408/b675f70a/attachment.html>

From thilo at ginkel.com  Wed Apr  8 22:01:57 2020
From: thilo at ginkel.com (Thilo-Alexander Ginkel)
Date: Thu, 9 Apr 2020 00:01:57 +0200
Subject: [Freeswitch-users] execute_on_answer equivalent for A-leg or
 bridged call?
In-Reply-To: <CAEJMVkDgcHRCHXOsNGC-b42wrQ1SOUK8SCxJ-m05su=vQmw9Rw@mail.gmail.com>
References: <CANvSZQ-fFJpk2UK4vhq4S6qP1k7fh-eYMeQA5eDkesNVesuCfA@mail.gmail.com>
 <CAEJMVkDgcHRCHXOsNGC-b42wrQ1SOUK8SCxJ-m05su=vQmw9Rw@mail.gmail.com>
Message-ID: <CANvSZQ82v1P9Y-JdqtA8s6p9wi1sr5YSH0V=cmHTOYCfVCme-w@mail.gmail.com>

On Wed, Apr 8, 2020 at 11:34 PM Brian West <brian at freeswitch.com> wrote:

> Set the variable on the a-leg
>

Thanks for your reply! I may have omitted an important detail: The A-leg
has already been answered before the bridge is initiated to obtain a
conference PIN via playAndGetDigits. So I assume that setting the variable
on the A-leg will never trigger anything on the A-leg, because it has
already been answered, right? Any alternatives?

Thanks,
Thilo
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200409/9068d0ff/attachment.html>

From david.villasmil.work at gmail.com  Wed Apr  8 22:20:36 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 8 Apr 2020 23:20:36 +0100
Subject: [Freeswitch-users] execute_on_answer equivalent for A-leg or
 bridged call?
In-Reply-To: <CANvSZQ-fFJpk2UK4vhq4S6qP1k7fh-eYMeQA5eDkesNVesuCfA@mail.gmail.com>
References: <CANvSZQ-fFJpk2UK4vhq4S6qP1k7fh-eYMeQA5eDkesNVesuCfA@mail.gmail.com>
Message-ID: <CAFGRPVrt0OwatkKiCihgrUmFkSyungEAtzTffXhkCO_UEPbAjg@mail.gmail.com>

I’m wondering maybe you can use play on the inbound uuid?

On Wed, 8 Apr 2020 at 21:41, Thilo-Alexander Ginkel <thilo at ginkel.com>
wrote:

> Hello everyone,
>
> I am attempting to play a sound on both legs after creating a bridge
> once the B-leg has answered. execute_on_answer explicitly only works
> for the newly answered channel, so no sound is playing on the
> (previously already connected) A-leg.
>
> Is there a way to make sure the sound is playing on both legs / the
> bridged call?
>
> ATM, I'm doing this (in Lua):
>
> session:execute("bridge", "{origination_callee_id_name='" .. callee_id
> .. "'}{execute_on_answer='playback
> tone_stream://%(500,0,500)'}user/jitsi@" .. domain_name)
>
> Thanks,
> Thilo
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200408/cb2147e6/attachment.html>

From vma at vallimamod.org  Thu Apr  9 08:04:26 2020
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Thu, 9 Apr 2020 10:04:26 +0200
Subject: [Freeswitch-users] execute_on_answer equivalent for A-leg or
 bridged call?
In-Reply-To: <CANvSZQ-fFJpk2UK4vhq4S6qP1k7fh-eYMeQA5eDkesNVesuCfA@mail.gmail.com>
References: <CANvSZQ-fFJpk2UK4vhq4S6qP1k7fh-eYMeQA5eDkesNVesuCfA@mail.gmail.com>
Message-ID: <46D222AF-B652-4EDF-AD81-4140C8F2B966@vallimamod.org>

Hi,

Have you tried uuid_broadcast? It has a "both" option which may help achieve what you want.


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


> On 8 Apr 2020, at 21:54, Thilo-Alexander Ginkel <thilo at ginkel.com> wrote:
> 
> Hello everyone,
> 
> I am attempting to play a sound on both legs after creating a bridge
> once the B-leg has answered. execute_on_answer explicitly only works
> for the newly answered channel, so no sound is playing on the
> (previously already connected) A-leg.
> 
> Is there a way to make sure the sound is playing on both legs / the
> bridged call?
> 
> ATM, I'm doing this (in Lua):
> 
> session:execute("bridge", "{origination_callee_id_name='" .. callee_id
> .. "'}{execute_on_answer='playback
> tone_stream://%(500,0,500)'}user/jitsi@" .. domain_name)
> 
> Thanks,
> Thilo
> 
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200409/20a6712b/attachment-0001.html>

From chad at apartmentlines.com  Thu Apr  9 17:41:29 2020
From: chad at apartmentlines.com (Chad Phillips)
Date: Thu, 9 Apr 2020 10:41:29 -0700
Subject: [Freeswitch-users] Using channel variables as values to time of day
	attributes in conditions
Message-ID: <CAJ0sghRVp+ipt-Y73y8X0GBL-duxwcTngGcsV5ayHqxcTV7w=Q@mail.gmail.com>

Will this dialplan XML condition work in newer versions of FreeSWITCH?

<condition wday="${REFORWARDING_DAYS}" hour="${REFORWARDING_HOURS}"
break="never">

${REFORWARDING_DAYS} and ${REFORWARDING_HOURS} are dialplan variables I set
prior to calling the extension with that condition. I tried this on an
older version of FreeSWITCH and it doesn't seem to work, I end up having to
hard code the values for the 'wday' and 'hour' attributes.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200409/07ecedab/attachment.html>

From brian at freeswitch.com  Thu Apr  9 19:14:28 2020
From: brian at freeswitch.com (Brian West)
Date: Thu, 9 Apr 2020 14:14:28 -0500
Subject: [Freeswitch-users] Using channel variables as values to time of
 day attributes in conditions
In-Reply-To: <CAJ0sghRVp+ipt-Y73y8X0GBL-duxwcTngGcsV5ayHqxcTV7w=Q@mail.gmail.com>
References: <CAJ0sghRVp+ipt-Y73y8X0GBL-duxwcTngGcsV5ayHqxcTV7w=Q@mail.gmail.com>
Message-ID: <CAEJMVkAHY6pSnKwMCtaD0JUsxmVYzTKjps1MOS8H3Q_+p_ZGXQ@mail.gmail.com>

I do not believe we do any sort of expansion on those fields.

On Thu, Apr 9, 2020 at 1:03 PM Chad Phillips <chad at apartmentlines.com>
wrote:

> Will this dialplan XML condition work in newer versions of FreeSWITCH?
>
> <condition wday="${REFORWARDING_DAYS}" hour="${REFORWARDING_HOURS}"
> break="never">
>
> ${REFORWARDING_DAYS} and ${REFORWARDING_HOURS} are dialplan variables I
> set prior to calling the extension with that condition. I tried this on an
> older version of FreeSWITCH and it doesn't seem to work, I end up having to
> hard code the values for the 'wday' and 'hour' attributes.
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 

Brian West | Co-founder and Developer

Need Commercial support? email sales at freeswitch.com

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

Email: brian at freeswitch.com

Mobile: 918-424-9378

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

[image: https://www.facebook.com/signalwireinc?src=email]
<https://www.facebook.com/freeswitch> [image:
https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200409/7e89e824/attachment.html>

From alexandru.tripon at itsyscom.com  Thu Apr  9 11:18:15 2020
From: alexandru.tripon at itsyscom.com (Alexandru Tripon)
Date: Thu, 9 Apr 2020 14:18:15 +0300
Subject: [Freeswitch-users] Accessing the Identity header in the dialplan
Message-ID: <CAAkCrRP+N0ZUGgGJf20N+AW=r8fu_CqjbrHpR05eB_VDnEMvGQ@mail.gmail.com>

Hello,

I wish to kindly ask how can I access the "Identity" header.
I set the following variable to true in my sip profile:`<param
name="parse-all-invite-headers" value="true"/>` in order to access all
invite header but without luck.
The SIP invite header is the following:
`
INVITE sip:1002 at 192.168.59.203 SIP/2.0.
Record-Route: <sip:192.168.59.203;lr>.
Call-ID: dd566f5f1c0c7f746226fdd374fdb2ae at 0:0:0:0:0:0:0:0.
CSeq: 1 INVITE.
From: "1001" <sip:1001 at 192.168.59.203>;tag=1bcc4483.
To: <sip:1002 at 192.168.59.203>.
Via: SIP/2.0/UDP 192.168.59.203:5060;branch=z9hG4bK2003.33316c51.0.
Via: SIP/2.0/UDP 192.168.59.213:5060
;branch=z9hG4bK-323138-463e2aa4a02196f588595ac028866af3.
Max-Forwards: 69.
Contact: "1001" <sip:1001 at 192.168.59.213:5060
;transport=udp;registering_acc=192_168_59_203>.
User-Agent: Jitsi2.11.20200408Linux.
Content-Type: application/sdp.
Content-Length: 897.
Date: Thu, 09 Apr 2020 10:11:24 GMT.
Identity:
eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiaHR0cHM6Ly9jZXJ0LmV4YW1wbGUub3JnL3Bhc3Nwb3J0LmNlciJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyI8bnVsbD4iXX0sImlhdCI6MTU4NjQyNzA4NCwib3JpZyI6eyJ0biI6IjxudWxsPiJ9LCJvcmlnaWQiOiIxMjM0NTYifQ.MEQCIH9KqTnrIW6pAlCqrYbHEFoCF3AsS6crh4ptSEzfnOM_AiAjZrZJ2Ur1csuYvJN8NygYEg1N-3RseLSoNHQOEX_Z5w;info=<
https://cert.example.org/passport.cer>;ppt="shaken".
Test-Header: mytest.
`
And the message log I get is :
`
variable_sip_i_from: ["1001" <sip:1001 at 192.168.59.203>;tag=1bcc4483]
variable_sip_i_to: [<sip:1002 at 192.168.59.203>]
variable_sip_i_call_id: [dd566f5f1c0c7f746226fdd374fdb2ae at 0:0:0:0:0:0:0:0]
variable_sip_i_cseq: [1 INVITE]
variable_sip_i_max_forwards: [69]
variable_sip_i_contact: ["1001" <sip:1001 at 192.168.59.213:5060
;transport=udp;registering_acc=192_168_59_203>]
variable_sip_i_user_agent: [Jitsi2.11.20200408Linux]
variable_sip_i_date: [Thu, 09 Apr 2020 10:11:24 GMT]
variable_sip_i_content_type: [application/sdp]
variable_sip_i_content_length: [897]
variable_sip_i_via: [ARRAY::SIP/2.0/UDP
192.168.59.203:5060;branch=z9hG4bK2003.33316c51.0;received=192.168.58.201|:SIP/2.0/UDP
192.168.59.213:5060;branch=z9hG4bK-323138-463e2aa4a02196f588595ac028866af3]
variable_sip_i_record_route: [<sip:192.168.59.203;lr>]
variable_sip_i_test_header: [mytest]
`
I tested this in latest release(
https://github.com/signalwire/freeswitch/releases/tag/v1.10.2) and on
master(commit: ae0444e9cbccdee55a80467d605e1e8c3363a36d) and the same thing
happened: the Test-Header is present in the log but the Identity header is
not.
How can I access the "Identity" header in the dialplan?

Thanks for your time,
Tripon Alexandru
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200409/8de4b8f0/attachment.html>

From Gilles at Sauvaire.com  Thu Apr  9 09:15:44 2020
From: Gilles at Sauvaire.com (Gilles SAUVAIRE)
Date: Thu, 9 Apr 2020 11:15:44 +0200
Subject: [Freeswitch-users] Mod_conference with many participants.
	Question for experts
In-Reply-To: <CAJs+-5rT64wgkzitkk6AnUuEpLMaGy271_LsWgapC8Q--1vNkw@mail.gmail.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAAAyq5Nx/9DRDrZ3hLNfSk2jCgAAAEAAAAH1u+aSboX1HjV4zgUKc+EcBAAAAAA==@Sauvaire.com>
 <CAJs+-5rT64wgkzitkk6AnUuEpLMaGy271_LsWgapC8Q--1vNkw@mail.gmail.com>
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAAAyq5Nx/9DRDrZ3hLNfSk2jCgAAAEAAAAF3mcMN/e+xDsjahHtcJvtoBAAAAAA==@Sauvaire.com>

Hello,

Sorry for my forgetfulness.

My OS is :
Debian 10.3
Linux xxxxx 4.19.0-8-amd64 #1 SMP Debian 4.19.98-1 (2020-01-26) x86_64 GNU/Linux

My freeswitch version :
freeswitch at xxxxx> version
FreeSWITCH Version 1.10.2-release+git~20191231T140119Z~f7bdd3845a~64bit (git f7bdd38 2019-12-31 14:01:19Z 64bit)

to do my tests I have a second freeswitch, I generate 96 calls to my conference freeswitch, with the first in normal conference and all the others in silent conference. 
the calls plays a voice message with a musical background.

then I make a call with my phone.
I hear the conference participant's plays message, but of very poor quality.

on the other hand if I change
<param name = "interval" value = "20" /> to  60 the same test works perfectly.

I can recreate the malfunction if you need tests or parameters during the malfunction.

I did a htop during the conference, but nothing special ...
the server is not 100%


what do you think of my conference profile ? is OK ?

on the other hand with the interval at 60 it works and the quality does not seem to be less good ?

what exactly does this interval setting change ?

big thanks to you for your help.

Gilles


De : FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Seven Du
Envoyé : lundi 6 avril 2020 02:50
À : FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Objet : Re: [Freeswitch-users] Mod_conference with many participants. Question for experts

You provided a lot of details but missed some important points. What OS? What version of FreeSWITCH? 

I have been running 600+ members across 5 servers even some with videos in one conference and not have any issue. Most members were muted.

On Tue, Mar 31, 2020 at 4:25 AM Gilles SAUVAIRE <mailto:Gilles at sauvaire.com> wrote:
Hello,

I am making a conference with 100 participants, on an 8-core server, 16
gigabytes of ram dedicated to my freeswitch.
98 participant are in mute mode and only two participant active.

8 times out of 10 the conference does not work. (very poor quality hatched
audio)
2 times out of 10 the conference works for a few minutes then becomes
unstable. (very poor quality hatched audio)

A htop at that time shows that all the cores are used but not more than 30%.
There is also plenty of memory available on the server.

    <profile name = "000">

      <param name = "domain" value = "$$ {domain}" />
      <param name = "rate" value = "8000" />
      <param name = "interval" value = "20" />
      <param name = "caller-controls" value = "none" />
      <param name = "moderator-controls" value = "none" />
      <param name = "moh-sound" value = "silence" />
      <param name = "suppress-events" value = "start-talking, stop-talking"
/>

      <! - Energy level required for audio to be sent to the other users ->
      <param name = "energy-level" value = "100" />
      <param name = "comfort-noise" value = "0" />
      <param name = "conference-flags" value = "audio-always" />

    </profile>

I have already found a solution, just put interval at 60.
      <param name = "interval" value = "60" />
With this the conference with 100 participants and only 2 active works.

I have many qustions

1)
It's strange, the conference with all the members in mute attribute,
shouldn't it normally do nothing?
Mute members should be transparent when it comes to mixing, is that just
broadcast?
( conference data = 66123 at 000++flags{endconf|nomoh|mute} )
Did I make a mistake in my profile? Is it normal ?


2)
Is there an easier way to broadcast?
That is, a call that "talks" to 90 calls?
My idea is to make a conference with 2 or 3 participants and to broadcast
this conference to all the other participants since he is not speaking.

many thanks to you for your advice and ideas.

Gilles





_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

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

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

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



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn



From david.villasmil.work at gmail.com  Fri Apr 10 01:23:28 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 10 Apr 2020 02:23:28 +0100
Subject: [Freeswitch-users] Mod_conference with many participants.
 Question for experts
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAAAyq5Nx/9DRDrZ3hLNfSk2jCgAAAEAAAAF3mcMN/e+xDsjahHtcJvtoBAAAAAA==@Sauvaire.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAAAyq5Nx/9DRDrZ3hLNfSk2jCgAAAEAAAAH1u+aSboX1HjV4zgUKc+EcBAAAAAA==@Sauvaire.com>
 <CAJs+-5rT64wgkzitkk6AnUuEpLMaGy271_LsWgapC8Q--1vNkw@mail.gmail.com>
 <!&!AAAAAAAAAAAYAAAAAAAAAAyq5Nx/9DRDrZ3hLNfSk2jCgAAAEAAAAF3mcMN/e+xDsjahHtcJvtoBAAAAAA==@Sauvaire.com>
Message-ID: <CAFGRPVoEZhHYKdWpA88N9rUk5hjsOYmNmdkTO0WPRRAocuJt9g@mail.gmail.com>

Interval is the ptime, the shorter, the more packets are sent out.

On Fri, 10 Apr 2020 at 02:17, Gilles SAUVAIRE <Gilles at sauvaire.com> wrote:

> Hello,
>
> Sorry for my forgetfulness.
>
> My OS is :
> Debian 10.3
> Linux xxxxx 4.19.0-8-amd64 #1 SMP Debian 4.19.98-1 (2020-01-26) x86_64
> GNU/Linux
>
> My freeswitch version :
> freeswitch at xxxxx> version
> FreeSWITCH Version 1.10.2-release+git~20191231T140119Z~f7bdd3845a~64bit
> (git f7bdd38 2019-12-31 14:01:19Z 64bit)
>
> to do my tests I have a second freeswitch, I generate 96 calls to my
> conference freeswitch, with the first in normal conference and all the
> others in silent conference.
> the calls plays a voice message with a musical background.
>
> then I make a call with my phone.
> I hear the conference participant's plays message, but of very poor
> quality.
>
> on the other hand if I change
> <param name = "interval" value = "20" /> to  60 the same test works
> perfectly.
>
> I can recreate the malfunction if you need tests or parameters during the
> malfunction.
>
> I did a htop during the conference, but nothing special ...
> the server is not 100%
>
>
> what do you think of my conference profile ? is OK ?
>
> on the other hand with the interval at 60 it works and the quality does
> not seem to be less good ?
>
> what exactly does this interval setting change ?
>
> big thanks to you for your help.
>
> Gilles
>
>
> De : FreeSWITCH-users [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] De la part de Seven Du
> Envoyé : lundi 6 avril 2020 02:50
> À : FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Objet : Re: [Freeswitch-users] Mod_conference with many participants.
> Question for experts
>
> You provided a lot of details but missed some important points. What OS?
> What version of FreeSWITCH?
>
> I have been running 600+ members across 5 servers even some with videos in
> one conference and not have any issue. Most members were muted.
>
> On Tue, Mar 31, 2020 at 4:25 AM Gilles SAUVAIRE <mailto:
> Gilles at sauvaire.com> wrote:
> Hello,
>
> I am making a conference with 100 participants, on an 8-core server, 16
> gigabytes of ram dedicated to my freeswitch.
> 98 participant are in mute mode and only two participant active.
>
> 8 times out of 10 the conference does not work. (very poor quality hatched
> audio)
> 2 times out of 10 the conference works for a few minutes then becomes
> unstable. (very poor quality hatched audio)
>
> A htop at that time shows that all the cores are used but not more than
> 30%.
> There is also plenty of memory available on the server.
>
>     <profile name = "000">
>
>       <param name = "domain" value = "$$ {domain}" />
>       <param name = "rate" value = "8000" />
>       <param name = "interval" value = "20" />
>       <param name = "caller-controls" value = "none" />
>       <param name = "moderator-controls" value = "none" />
>       <param name = "moh-sound" value = "silence" />
>       <param name = "suppress-events" value = "start-talking, stop-talking"
> />
>
>       <! - Energy level required for audio to be sent to the other users ->
>       <param name = "energy-level" value = "100" />
>       <param name = "comfort-noise" value = "0" />
>       <param name = "conference-flags" value = "audio-always" />
>
>     </profile>
>
> I have already found a solution, just put interval at 60.
>       <param name = "interval" value = "60" />
> With this the conference with 100 participants and only 2 active works.
>
> I have many qustions
>
> 1)
> It's strange, the conference with all the members in mute attribute,
> shouldn't it normally do nothing?
> Mute members should be transparent when it comes to mixing, is that just
> broadcast?
> ( conference data = 66123 at 000++flags{endconf|nomoh|mute} )
> Did I make a mistake in my profile? Is it normal ?
>
>
> 2)
> Is there an easier way to broadcast?
> That is, a call that "talks" to 90 calls?
> My idea is to make a conference with 2 or 3 participants and to broadcast
> this conference to all the other participants since he is not speaking.
>
> many thanks to you for your advice and ideas.
>
> Gilles
>
>
>
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> mailto:sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> mailto:FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
>
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200410/2826ee06/attachment.html>

From david.villasmil.work at gmail.com  Fri Apr 10 20:15:52 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 10 Apr 2020 21:15:52 +0100
Subject: [Freeswitch-users] hangup_hook on an originated leg
Message-ID: <CAFGRPVp7PHtsOBi9chyEJMu2C3Vsf0wMWnveNfyytC-23Uf=Lg@mail.gmail.com>

Hello,

So I'm receiving an INVITE that i'm simply use to generate an origination,
something like:

<extension name="in_out_test">
    <condition field="destination_number" expression="^(whatever)$">
            <action application="log" data="ERR Starting lua script..."/>
            <action application="info" data=""/>
            <action application="set" data="hangup_after_bridge=true"/>
            <action application="python" data="foo-outbound"/>
    </condition>
</extension>

With that in a python script in which i need to set a hangup_hook (I need
the originated variables like billsec, etc) like this:

# Some headers data gathering, then:
session.execute( "set", "session_in_hangup_hook=true" )
session.execute( "export", "nolocal:api_hangup_hook=python cdr" )
call_command = '{cldnum=' + CallStruct['to'] + \
    ',ignore_early_media=true,audio1=' + CallStruct['audios'] + \
    ',app_campaing_type=' + CallStruct['type'] + \
    ',originate_retries=' + CallStruct['retry'] + \
    ',app_phone_desc=' + CallStruct['phone_desc'] + \
    ',app_gateway=' + CallStruct['gateway'] + \
    ',originate_timeout=30' + \
    ',app_from=' + CallStruct['to'] + \
    ',app_to=' + CallStruct['from'] + \
    ',origination_callee_id_number=' + CallStruct['to'] + \
    ',origination_caller_id_number=' + CallStruct['from'] + \
    '}sofia/gateway/' + CallStruct['gateway'] + '/' + CallStruct['to'] + '
9999 XML public'

But this doesn't seem to be doing anything..

Anyone ever done this before?

Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200410/89c96195/attachment.html>

From david.villasmil.work at gmail.com  Fri Apr 10 20:29:31 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 10 Apr 2020 21:29:31 +0100
Subject: [Freeswitch-users] hangup_hook on an originated leg
In-Reply-To: <CAFGRPVp7PHtsOBi9chyEJMu2C3Vsf0wMWnveNfyytC-23Uf=Lg@mail.gmail.com>
References: <CAFGRPVp7PHtsOBi9chyEJMu2C3Vsf0wMWnveNfyytC-23Uf=Lg@mail.gmail.com>
Message-ID: <CAFGRPVo5u+tgp4YBfuOToq8MT+Ur5a33O1B-Fr3YFomvHrspZA@mail.gmail.com>

Ok, so to answer myself, since on an origination there isn't a- and b-leg,
there's no need to export...
So this is what works:

    session.execute( "set", "session_in_hangup_hook=true" )
    session.execute( "set", "api_hangup_hook=python cdr" )


Thanks everyone!

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337


On Fri, Apr 10, 2020 at 9:15 PM David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello,
>
> So I'm receiving an INVITE that i'm simply use to generate an origination,
> something like:
>
> <extension name="in_out_test">
>     <condition field="destination_number" expression="^(whatever)$">
>             <action application="log" data="ERR Starting lua script..."/>
>             <action application="info" data=""/>
>             <action application="set" data="hangup_after_bridge=true"/>
>             <action application="python" data="foo-outbound"/>
>     </condition>
> </extension>
>
> With that in a python script in which i need to set a hangup_hook (I need
> the originated variables like billsec, etc) like this:
>
> # Some headers data gathering, then:
> session.execute( "set", "session_in_hangup_hook=true" )
> session.execute( "export", "nolocal:api_hangup_hook=python cdr" )
> call_command = '{cldnum=' + CallStruct['to'] + \
>     ',ignore_early_media=true,audio1=' + CallStruct['audios'] + \
>     ',app_campaing_type=' + CallStruct['type'] + \
>     ',originate_retries=' + CallStruct['retry'] + \
>     ',app_phone_desc=' + CallStruct['phone_desc'] + \
>     ',app_gateway=' + CallStruct['gateway'] + \
>     ',originate_timeout=30' + \
>     ',app_from=' + CallStruct['to'] + \
>     ',app_to=' + CallStruct['from'] + \
>     ',origination_callee_id_number=' + CallStruct['to'] + \
>     ',origination_caller_id_number=' + CallStruct['from'] + \
>     '}sofia/gateway/' + CallStruct['gateway'] + '/' + CallStruct['to'] + '
> 9999 XML public'
>
> But this doesn't seem to be doing anything..
>
> Anyone ever done this before?
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200410/790febb6/attachment.html>

From josefu at gmail.com  Mon Apr 13 09:49:14 2020
From: josefu at gmail.com (=?UTF-8?Q?Jose_Fco=2E_Irles_Dur=C3=A1?=)
Date: Mon, 13 Apr 2020 11:49:14 +0200
Subject: [Freeswitch-users] Answer ringing phone with uuid_phone_event
	behind load balancer
Message-ID: <CAJOPU6VHjef1X9dXDsJ8=sfAK_iK4VjG3Yt95TKPQAL6tWfYKQ@mail.gmail.com>

Hi,

I'm testing the command "uuid_phone_event <uuid> talk" for answer
ringing channels on remote phones.

With a standalone configuration this works perfectly, the NOTIFY with
the "Event: talk" is sended to the phone.

The problem comes when I have a load balancer (for managing
registers), the request uri of the NOTIFY is populated with the value
of "sofia_contact" function (I think, because is the value that I
write for this function via xml_curl).

I think that the request URI for this message must be the contact
value for this dialog and not the "sofia_contact" value.

I'm right?

Best regards

--
Jose Fco. Irles Durá


From mrjoli021 at gmail.com  Tue Apr 14 16:20:13 2020
From: mrjoli021 at gmail.com (Joli Martinez)
Date: Tue, 14 Apr 2020 12:20:13 -0400
Subject: [Freeswitch-users] Changing RTP ports mid call
Message-ID: <CANeteid2sAAGfeFjFw3uq0GTLdjMooD1tou6gMJsu309GeWOEg@mail.gmail.com>

Hello,

We have a phone app that ties back to our freeswitch servers.  Recently the
android phones are getting one-way audio.  It works on iPhones and desktop
clients.  Based on the pcap's and fs_cli I see that the audio ports changed
after the sdp negotiations.  Not sure why this is happening also see that
the codec is set to "BUG Codec PCMU:0"

Any Suggestions?

2020-04-14 15:56:07.661344 [DEBUG] switch_ivr_async.c:1500 No silence
detection configured; assuming start of speech
2020-04-14 15:56:07.701439 [DEBUG] switch_core_io.c:448 Setting BUG Codec
PCMU:0
2020-04-14 15:56:07.701439 [DEBUG] switch_rtp.c:1887 rtcp_stats_init: audio
ssrc[1321291818] base_seq[30216]
2020-04-14 15:56:07.881347 [INFO] switch_rtp.c:7231 Auto Changing audio
port from 124.123.104.212:10488 to 124.123.104.212:9658

Thanks,
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200414/269ebc91/attachment.html>

From mickael.bride at orange.com  Fri Apr 10 15:37:16 2020
From: mickael.bride at orange.com (mickael.bride at orange.com)
Date: Fri, 10 Apr 2020 15:37:16 +0000
Subject: [Freeswitch-users] FreeSwitch for WebRTC to SIP/RTP gateway?
Message-ID: <1568_1586533037_5E9092AD_1568_145_1_171AD92A086E0B4DB629A5710DBA738137BB64DB@OPEXCNORM53.corporate.adroot.infra.ftgroup>

Hello all,

Sorry in advance if my email is not clear enough or relevant.
My need is to have some kind of WebRTC gateway between a web application and a call server managing SIP/RTP (the webapp is a softphone, audio only).
This call server is not exposed on Internet, this is mainly why I need a proxy/gateway.

I am quite a newbie on WebRTC / SIP / RTP.
>From what I have seen on Internet some projects/products are doing that kind of functionality (Asterisk, Doubango webrtc2sip, Janus, Audio Code ...) , but I am not able to focus on the simplest way to do it. Also some are commercial products but I would prefer open source, installed on CentOS.
FreeSwitch seems really great but I wonder if it is not too "big" or complicated for my needs.

Questions are:

·         Is FreeSwitch relevant to have a proxy/gateway between a WebRTC web app and a call server not exposed on Internet?

·         Is it quite "simple" to install and configure freeSwitch to have that functionality?

·         Is there any other simpler project to do this?

Any advice would be really appreciate.

Thank you,

Mickaël


_________________________________________________________________________________________________________________________

Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc
pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler
a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration,
Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci.

This message and its attachments may contain confidential or privileged information that may be protected by law;
they should not be distributed, used or copied without authorisation.
If you have received this email in error, please notify the sender and delete this message and its attachments.
As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified.
Thank you.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200410/f25c3ba5/attachment.html>

From svanherwaarden at precisionag.org  Fri Apr 10 19:40:32 2020
From: svanherwaarden at precisionag.org (Sam van Herwaarden)
Date: Fri, 10 Apr 2020 21:40:32 +0200
Subject: [Freeswitch-users] Rejecting calls (for a missed call service)
Message-ID: <CAAtWTemn9TeVsFu3q_TXStBfe2Q59ThwOjOyiPmz_fSEdZsfLQ@mail.gmail.com>

Hi,

I am working on an IVR system that is supposed to work as a missed call
service: users call, it rings for a moment, either they or the system hangs
up, and then the system places a call to them. The goal is to avoid charges
on their end.

I basically have this working but it doesn’t sound natural on the user’s
side. It doesn’t sound like the phone is ringing, there are just two beeps
before it disconnects. I am using the ‘hangup’ command, I’ve tried the
default and using a CALL_REJECTED status. The latter is worse - I get a
voice from the telco saying “the number you are trying to reach is
unavailable” (not at all what I want). This is in India for what it’s worth.

Does anyone have experience setting up such a system and advice on the
commands I should be using?

Thanks!

Kind regards,
Sam
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200410/d6482bf2/attachment.html>

From gmaruzz at gmail.com  Wed Apr 15 08:48:30 2020
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 15 Apr 2020 10:48:30 +0200
Subject: [Freeswitch-users] FreeSwitch for WebRTC to SIP/RTP gateway?
In-Reply-To: <1568_1586533037_5E9092AD_1568_145_1_171AD92A086E0B4DB629A5710DBA738137BB64DB@OPEXCNORM53.corporate.adroot.infra.ftgroup>
References: <1568_1586533037_5E9092AD_1568_145_1_171AD92A086E0B4DB629A5710DBA738137BB64DB@OPEXCNORM53.corporate.adroot.infra.ftgroup>
Message-ID: <CALXCt0osa56f_FK=1kTUTcWaaT8ACb=Zt6gvZL=sOs4FD61n6A@mail.gmail.com>

Yes, is relevant
Yes, is simple: after a standard install, modify internal.xml SIP profile
and activate port 7443 as wss port, then write dialplan that bridge
incoming calls to outbound calls on your other sip server
You will need "real" ssl certificates, eg letsencrypt, from which you will
build the certificate needed by freeswitch for webrtc:

cat /etc/dehydrated/certs/XXX/fullchain.pem
/etc/dehydrated/certs/XXX/privkey.pem > /etc/freeswitch/tls/wss.pem

That's it

-giovanni


On Wed, Apr 15, 2020 at 12:45 AM <mickael.bride at orange.com> wrote:

> Hello all,
>
>
>
> Sorry in advance if my email is not clear enough or relevant.
>
> My need is to have some kind of WebRTC gateway between a web application
> and a call server managing SIP/RTP (the webapp is a softphone, audio only).
>
> This call server is not exposed on Internet, this is mainly why I need a
> proxy/gateway.
>
>
>
> I am quite a newbie on WebRTC / SIP / RTP.
>
> From what I have seen on Internet some projects/products are doing that
> kind of functionality (Asterisk, Doubango webrtc2sip, Janus, Audio Code …)
> , but I am not able to focus on the simplest way to do it. Also some are
> commercial products but I would prefer open source, installed on CentOS.
>
> FreeSwitch seems really great but I wonder if it is not too “big” or
> complicated for my needs.
>
>
>
> Questions are:
>
> ·         Is FreeSwitch relevant to have a proxy/gateway between a WebRTC
> web app and a call server not exposed on Internet?
>
> ·         Is it quite “simple” to install and configure freeSwitch to
> have that functionality?
>
> ·         Is there any other simpler project to do this?
>
>
>
> Any advice would be really appreciate.
>
>
>
> Thank you,
>
>
>
> Mickaël
>
>
>
> _________________________________________________________________________________________________________________________
>
> Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc
> pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler
> a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration,
> Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci.
>
> This message and its attachments may contain confidential or privileged information that may be protected by law;
> they should not be distributed, used or copied without authorisation.
> If you have received this email in error, please notify the sender and delete this message and its attachments.
> As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified.
> Thank you.
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200415/6f2a778c/attachment-0001.html>

From covici at ccs.covici.com  Wed Apr 15 10:39:37 2020
From: covici at ccs.covici.com (John Covici)
Date: Wed, 15 Apr 2020 06:39:37 -0400
Subject: [Freeswitch-users] FreeSwitch for WebRTC to SIP/RTP gateway?
In-Reply-To: <CALXCt0osa56f_FK=1kTUTcWaaT8ACb=Zt6gvZL=sOs4FD61n6A@mail.gmail.com>
References: <1568_1586533037_5E9092AD_1568_145_1_171AD92A086E0B4DB629A5710DBA738137BB64DB@OPEXCNORM53.corporate.adroot.infra.ftgroup>
 <CALXCt0osa56f_FK=1kTUTcWaaT8ACb=Zt6gvZL=sOs4FD61n6A@mail.gmail.com>
Message-ID: <m3blntvxqu.wl-covici@ccs.covici.com>

You don't have the cert, just the full chain and private key -- is
that correct?

On Wed, 15 Apr 2020 04:48:30 -0400,
Giovanni Maruzzelli wrote:
> 
> [1  <multipart/alternative (7bit)>]
> [1.1  <text/plain; UTF-8 (quoted-printable)>]
> Yes, is relevant
> Yes, is simple: after a standard install, modify internal.xml SIP profile
> and activate port 7443 as wss port, then write dialplan that bridge
> incoming calls to outbound calls on your other sip server
> You will need "real" ssl certificates, eg letsencrypt, from which you will
> build the certificate needed by freeswitch for webrtc:
> 
> cat /etc/dehydrated/certs/XXX/fullchain.pem
> /etc/dehydrated/certs/XXX/privkey.pem > /etc/freeswitch/tls/wss.pem
> 
> That's it
> 
> -giovanni
> 
> 
> On Wed, Apr 15, 2020 at 12:45 AM <mickael.bride at orange.com> wrote:
> 
> > Hello all,
> >
> >
> >
> > Sorry in advance if my email is not clear enough or relevant.
> >
> > My need is to have some kind of WebRTC gateway between a web application
> > and a call server managing SIP/RTP (the webapp is a softphone, audio only).
> >
> > This call server is not exposed on Internet, this is mainly why I need a
> > proxy/gateway.
> >
> >
> >
> > I am quite a newbie on WebRTC / SIP / RTP.
> >
> > From what I have seen on Internet some projects/products are doing that
> > kind of functionality (Asterisk, Doubango webrtc2sip, Janus, Audio Code …)
> > , but I am not able to focus on the simplest way to do it. Also some are
> > commercial products but I would prefer open source, installed on CentOS.
> >
> > FreeSwitch seems really great but I wonder if it is not too “big” or
> > complicated for my needs.
> >
> >
> >
> > Questions are:
> >
> > ·         Is FreeSwitch relevant to have a proxy/gateway between a WebRTC
> > web app and a call server not exposed on Internet?
> >
> > ·         Is it quite “simple” to install and configure freeSwitch to
> > have that functionality?
> >
> > ·         Is there any other simpler project to do this?
> >
> >
> >
> > Any advice would be really appreciate.
> >
> >
> >
> > Thank you,
> >
> >
> >
> > Mickaël
> >
> >
> >
> > _________________________________________________________________________________________________________________________
> >
> > Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc
> > pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler
> > a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration,
> > Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci.
> >
> > This message and its attachments may contain confidential or privileged information that may be protected by law;
> > they should not be distributed, used or copied without authorisation.
> > If you have received this email in error, please notify the sender and delete this message and its attachments.
> > As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified.
> > Thank you.
> >
> > _________________________________________________________________________
> >
> > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> > services.
> > Build your next product on our scalable cloud platform.
> >
> > Join our online community to chat in real time
> > https://signalwire.community
> >
> > Professional FreeSWITCH Services
> > sales at freeswitch.com
> > https://freeswitch.com
> >
> > Official FreeSWITCH Sites
> > https://freeswitch.com/oss
> > https://freeswitch.org/confluence
> > https://cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > https://freeswitch.com
> 
> 
> 
> -- 
> Sincerely,
> 
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
> [1.2  <text/html; UTF-8 (quoted-printable)>]
> [2  <text/plain; utf-8 (base64)>]
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici wb2una
         covici at ccs.covici.com


From gmaruzz at gmail.com  Wed Apr 15 10:43:31 2020
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 15 Apr 2020 12:43:31 +0200
Subject: [Freeswitch-users] FreeSwitch for WebRTC to SIP/RTP gateway?
In-Reply-To: <m3blntvxqu.wl-covici@ccs.covici.com>
References: <1568_1586533037_5E9092AD_1568_145_1_171AD92A086E0B4DB629A5710DBA738137BB64DB@OPEXCNORM53.corporate.adroot.infra.ftgroup>
 <CALXCt0osa56f_FK=1kTUTcWaaT8ACb=Zt6gvZL=sOs4FD61n6A@mail.gmail.com>
 <m3blntvxqu.wl-covici@ccs.covici.com>
Message-ID: <CALXCt0qiyg6UJuDsSedug2Ec1vXes-2z4f=1B2rAkabopaZA2w@mail.gmail.com>

letsencrypt terminology: in fullchain.pem is comprised cert and CA


On Wed, Apr 15, 2020 at 12:39 PM John Covici <covici at ccs.covici.com> wrote:

> You don't have the cert, just the full chain and private key -- is
> that correct?
>
> On Wed, 15 Apr 2020 04:48:30 -0400,
> Giovanni Maruzzelli wrote:
> >
> > [1  <multipart/alternative (7bit)>]
> > [1.1  <text/plain; UTF-8 (quoted-printable)>]
> > Yes, is relevant
> > Yes, is simple: after a standard install, modify internal.xml SIP profile
> > and activate port 7443 as wss port, then write dialplan that bridge
> > incoming calls to outbound calls on your other sip server
> > You will need "real" ssl certificates, eg letsencrypt, from which you
> will
> > build the certificate needed by freeswitch for webrtc:
> >
> > cat /etc/dehydrated/certs/XXX/fullchain.pem
> > /etc/dehydrated/certs/XXX/privkey.pem > /etc/freeswitch/tls/wss.pem
> >
> > That's it
> >
> > -giovanni
> >
> >
> > On Wed, Apr 15, 2020 at 12:45 AM <mickael.bride at orange.com> wrote:
> >
> > > Hello all,
> > >
> > >
> > >
> > > Sorry in advance if my email is not clear enough or relevant.
> > >
> > > My need is to have some kind of WebRTC gateway between a web
> application
> > > and a call server managing SIP/RTP (the webapp is a softphone, audio
> only).
> > >
> > > This call server is not exposed on Internet, this is mainly why I need
> a
> > > proxy/gateway.
> > >
> > >
> > >
> > > I am quite a newbie on WebRTC / SIP / RTP.
> > >
> > > From what I have seen on Internet some projects/products are doing that
> > > kind of functionality (Asterisk, Doubango webrtc2sip, Janus, Audio
> Code …)
> > > , but I am not able to focus on the simplest way to do it. Also some
> are
> > > commercial products but I would prefer open source, installed on
> CentOS.
> > >
> > > FreeSwitch seems really great but I wonder if it is not too “big” or
> > > complicated for my needs.
> > >
> > >
> > >
> > > Questions are:
> > >
> > > ·         Is FreeSwitch relevant to have a proxy/gateway between a
> WebRTC
> > > web app and a call server not exposed on Internet?
> > >
> > > ·         Is it quite “simple” to install and configure freeSwitch to
> > > have that functionality?
> > >
> > > ·         Is there any other simpler project to do this?
> > >
> > >
> > >
> > > Any advice would be really appreciate.
> > >
> > >
> > >
> > > Thank you,
> > >
> > >
> > >
> > > Mickaël
> > >
> > >
> > >
> > >
> _________________________________________________________________________________________________________________________
> > >
> > > Ce message et ses pieces jointes peuvent contenir des informations
> confidentielles ou privilegiees et ne doivent donc
> > > pas etre diffuses, exploites ou copies sans autorisation. Si vous avez
> recu ce message par erreur, veuillez le signaler
> > > a l'expediteur et le detruire ainsi que les pieces jointes. Les
> messages electroniques etant susceptibles d'alteration,
> > > Orange decline toute responsabilite si ce message a ete altere,
> deforme ou falsifie. Merci.
> > >
> > > This message and its attachments may contain confidential or
> privileged information that may be protected by law;
> > > they should not be distributed, used or copied without authorisation.
> > > If you have received this email in error, please notify the sender and
> delete this message and its attachments.
> > > As emails may be altered, Orange is not liable for messages that have
> been modified, changed or falsified.
> > > Thank you.
> > >
> > >
> _________________________________________________________________________
> > >
> > > The FreeSWITCH project is sponsored by SignalWire
> https://signalwire.com
> > > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> > > services.
> > > Build your next product on our scalable cloud platform.
> > >
> > > Join our online community to chat in real time
> > > https://signalwire.community
> > >
> > > Professional FreeSWITCH Services
> > > sales at freeswitch.com
> > > https://freeswitch.com
> > >
> > > Official FreeSWITCH Sites
> > > https://freeswitch.com/oss
> > > https://freeswitch.org/confluence
> > > https://cluecon.com
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > https://freeswitch.com
> >
> >
> >
> > --
> > Sincerely,
> >
> > Giovanni Maruzzelli
> > OpenTelecom.IT
> > cell: +39 347 266 56 18
> > [1.2  <text/html; UTF-8 (quoted-printable)>]
> > [2  <text/plain; utf-8 (base64)>]
> > _________________________________________________________________________
> >
> > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> > Build your next product on our scalable cloud platform.
> >
> > Join our online community to chat in real time
> https://signalwire.community
> >
> > Professional FreeSWITCH Services
> > sales at freeswitch.com
> > https://freeswitch.com
> >
> > Official FreeSWITCH Sites
> > https://freeswitch.com/oss
> > https://freeswitch.org/confluence
> > https://cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > https://freeswitch.com
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>          John Covici wb2una
>          covici at ccs.covici.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200415/a3756930/attachment.html>

From covici at ccs.covici.com  Wed Apr 15 12:24:41 2020
From: covici at ccs.covici.com (John Covici)
Date: Wed, 15 Apr 2020 08:24:41 -0400
Subject: [Freeswitch-users] FreeSwitch for WebRTC to SIP/RTP gateway?
In-Reply-To: <CALXCt0qiyg6UJuDsSedug2Ec1vXes-2z4f=1B2rAkabopaZA2w@mail.gmail.com>
References: <1568_1586533037_5E9092AD_1568_145_1_171AD92A086E0B4DB629A5710DBA738137BB64DB@OPEXCNORM53.corporate.adroot.infra.ftgroup>
 <CALXCt0osa56f_FK=1kTUTcWaaT8ACb=Zt6gvZL=sOs4FD61n6A@mail.gmail.com>
 <m3blntvxqu.wl-covici@ccs.covici.com>
 <CALXCt0qiyg6UJuDsSedug2Ec1vXes-2z4f=1B2rAkabopaZA2w@mail.gmail.com>
Message-ID: <m3a73dvsvq.wl-covici@ccs.covici.com>

OK, thanks.

On Wed, 15 Apr 2020 06:43:31 -0400,
Giovanni Maruzzelli wrote:
> 
> [1  <multipart/alternative (7bit)>]
> [1.1  <text/plain; UTF-8 (quoted-printable)>]
> letsencrypt terminology: in fullchain.pem is comprised cert and CA
> 
> 
> On Wed, Apr 15, 2020 at 12:39 PM John Covici <covici at ccs.covici.com> wrote:
> 
> > You don't have the cert, just the full chain and private key -- is
> > that correct?
> >
> > On Wed, 15 Apr 2020 04:48:30 -0400,
> > Giovanni Maruzzelli wrote:
> > >
> > > [1  <multipart/alternative (7bit)>]
> > > [1.1  <text/plain; UTF-8 (quoted-printable)>]
> > > Yes, is relevant
> > > Yes, is simple: after a standard install, modify internal.xml SIP profile
> > > and activate port 7443 as wss port, then write dialplan that bridge
> > > incoming calls to outbound calls on your other sip server
> > > You will need "real" ssl certificates, eg letsencrypt, from which you
> > will
> > > build the certificate needed by freeswitch for webrtc:
> > >
> > > cat /etc/dehydrated/certs/XXX/fullchain.pem
> > > /etc/dehydrated/certs/XXX/privkey.pem > /etc/freeswitch/tls/wss.pem
> > >
> > > That's it
> > >
> > > -giovanni
> > >
> > >
> > > On Wed, Apr 15, 2020 at 12:45 AM <mickael.bride at orange.com> wrote:
> > >
> > > > Hello all,
> > > >
> > > >
> > > >
> > > > Sorry in advance if my email is not clear enough or relevant.
> > > >
> > > > My need is to have some kind of WebRTC gateway between a web
> > application
> > > > and a call server managing SIP/RTP (the webapp is a softphone, audio
> > only).
> > > >
> > > > This call server is not exposed on Internet, this is mainly why I need
> > a
> > > > proxy/gateway.
> > > >
> > > >
> > > >
> > > > I am quite a newbie on WebRTC / SIP / RTP.
> > > >
> > > > From what I have seen on Internet some projects/products are doing that
> > > > kind of functionality (Asterisk, Doubango webrtc2sip, Janus, Audio
> > Code …)
> > > > , but I am not able to focus on the simplest way to do it. Also some
> > are
> > > > commercial products but I would prefer open source, installed on
> > CentOS.
> > > >
> > > > FreeSwitch seems really great but I wonder if it is not too “big” or
> > > > complicated for my needs.
> > > >
> > > >
> > > >
> > > > Questions are:
> > > >
> > > > ·         Is FreeSwitch relevant to have a proxy/gateway between a
> > WebRTC
> > > > web app and a call server not exposed on Internet?
> > > >
> > > > ·         Is it quite “simple” to install and configure freeSwitch to
> > > > have that functionality?
> > > >
> > > > ·         Is there any other simpler project to do this?
> > > >
> > > >
> > > >
> > > > Any advice would be really appreciate.
> > > >
> > > >
> > > >
> > > > Thank you,
> > > >
> > > >
> > > >
> > > > Mickaël
> > > >
> > > >
> > > >
> > > >
> > _________________________________________________________________________________________________________________________
> > > >
> > > > Ce message et ses pieces jointes peuvent contenir des informations
> > confidentielles ou privilegiees et ne doivent donc
> > > > pas etre diffuses, exploites ou copies sans autorisation. Si vous avez
> > recu ce message par erreur, veuillez le signaler
> > > > a l'expediteur et le detruire ainsi que les pieces jointes. Les
> > messages electroniques etant susceptibles d'alteration,
> > > > Orange decline toute responsabilite si ce message a ete altere,
> > deforme ou falsifie. Merci.
> > > >
> > > > This message and its attachments may contain confidential or
> > privileged information that may be protected by law;
> > > > they should not be distributed, used or copied without authorisation.
> > > > If you have received this email in error, please notify the sender and
> > delete this message and its attachments.
> > > > As emails may be altered, Orange is not liable for messages that have
> > been modified, changed or falsified.
> > > > Thank you.
> > > >
> > > >
> > _________________________________________________________________________
> > > >
> > > > The FreeSWITCH project is sponsored by SignalWire
> > https://signalwire.com
> > > > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> > > > services.
> > > > Build your next product on our scalable cloud platform.
> > > >
> > > > Join our online community to chat in real time
> > > > https://signalwire.community
> > > >
> > > > Professional FreeSWITCH Services
> > > > sales at freeswitch.com
> > > > https://freeswitch.com
> > > >
> > > > Official FreeSWITCH Sites
> > > > https://freeswitch.com/oss
> > > > https://freeswitch.org/confluence
> > > > https://cluecon.com
> > > >
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > https://freeswitch.com
> > >
> > >
> > >
> > > --
> > > Sincerely,
> > >
> > > Giovanni Maruzzelli
> > > OpenTelecom.IT
> > > cell: +39 347 266 56 18
> > > [1.2  <text/html; UTF-8 (quoted-printable)>]
> > > [2  <text/plain; utf-8 (base64)>]
> > > _________________________________________________________________________
> > >
> > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> > > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> > services.
> > > Build your next product on our scalable cloud platform.
> > >
> > > Join our online community to chat in real time
> > https://signalwire.community
> > >
> > > Professional FreeSWITCH Services
> > > sales at freeswitch.com
> > > https://freeswitch.com
> > >
> > > Official FreeSWITCH Sites
> > > https://freeswitch.com/oss
> > > https://freeswitch.org/confluence
> > > https://cluecon.com
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > https://freeswitch.com
> >
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> >
> >          John Covici wb2una
> >          covici at ccs.covici.com
> >
> > _________________________________________________________________________
> >
> > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> > services.
> > Build your next product on our scalable cloud platform.
> >
> > Join our online community to chat in real time
> > https://signalwire.community
> >
> > Professional FreeSWITCH Services
> > sales at freeswitch.com
> > https://freeswitch.com
> >
> > Official FreeSWITCH Sites
> > https://freeswitch.com/oss
> > https://freeswitch.org/confluence
> > https://cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > https://freeswitch.com
> 
> 
> 
> -- 
> Sincerely,
> 
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
> [1.2  <text/html; UTF-8 (quoted-printable)>]
> [2  <text/plain; utf-8 (base64)>]
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici wb2una
         covici at ccs.covici.com


From gmaruzz at gmail.com  Wed Apr 15 15:46:35 2020
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 15 Apr 2020 17:46:35 +0200
Subject: [Freeswitch-users] FreeSwitch for WebRTC to SIP/RTP gateway?
In-Reply-To: <926519743.991603.1586965101525@mail.yahoo.com>
References: <926519743.991603.1586965101525.ref@mail.yahoo.com>
 <926519743.991603.1586965101525@mail.yahoo.com>
Message-ID: <CALXCt0r2kqY9QFjzBmmFUg_tvV_C506-oKzbW9Nt6_3+SQuikw@mail.gmail.com>

at the bottom of the mail you find instruction on how to unsubscribe




On Wed, Apr 15, 2020 at 5:38 PM Yan Zhang <yan at petatrading.com> wrote:

> WHY I AM STILL RECEIVING THIS ENDLESS MAIL NOTICE?
>
> Please remove me. Thanks
>
>
>
> On Tuesday, April 14, 2020, 06:48:26 PM EDT, <mickael.bride at orange.com>
> wrote:
>
>
> Hello all,
>
>
>
> Sorry in advance if my email is not clear enough or relevant.
>
> My need is to have some kind of WebRTC gateway between a web application
> and a call server managing SIP/RTP (the webapp is a softphone, audio only).
>
> This call server is not exposed on Internet, this is mainly why I need a
> proxy/gateway.
>
>
>
> I am quite a newbie on WebRTC / SIP / RTP.
>
> From what I have seen on Internet some projects/products are doing that
> kind of functionality (Asterisk, Doubango webrtc2sip, Janus, Audio Code …)
> , but I am not able to focus on the simplest way to do it. Also some are
> commercial products but I would prefer open source, installed on CentOS.
>
> FreeSwitch seems really great but I wonder if it is not too “big” or
> complicated for my needs.
>
>
>
> Questions are:
>
> ·         Is FreeSwitch relevant to have a proxy/gateway between a WebRTC
> web app and a call server not exposed on Internet?
>
> ·         Is it quite “simple” to install and configure freeSwitch to
> have that functionality?
>
> ·         Is there any other simpler project to do this?
>
>
>
> Any advice would be really appreciate.
>
>
>
> Thank you,
>
>
>
> Mickaël
>
>
>
> _________________________________________________________________________________________________________________________
>
> Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc
> pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler
> a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration,
> Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci.
>
> This message and its attachments may contain confidential or privileged information that may be protected by law;
> they should not be distributed, used or copied without authorisation.
> If you have received this email in error, please notify the sender and delete this message and its attachments.
> As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified.
> Thank you.
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>


-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200415/cb457d0d/attachment.html>

From mitul at enterux.com  Wed Apr 15 16:08:41 2020
From: mitul at enterux.com (Mitul Limbani)
Date: Wed, 15 Apr 2020 21:38:41 +0530
Subject: [Freeswitch-users] FreeSwitch for WebRTC to SIP/RTP gateway?
In-Reply-To: <CALXCt0r2kqY9QFjzBmmFUg_tvV_C506-oKzbW9Nt6_3+SQuikw@mail.gmail.com>
References: <926519743.991603.1586965101525.ref@mail.yahoo.com>
 <926519743.991603.1586965101525@mail.yahoo.com>
 <CALXCt0r2kqY9QFjzBmmFUg_tvV_C506-oKzbW9Nt6_3+SQuikw@mail.gmail.com>
Message-ID: <CAAoGpGSpjkG=geRYNfpSXui1ai=JwS3pMW6cE_Cg+vjYN0oucw@mail.gmail.com>

Someone please do him a favor by kicking out

ML

On Wed, 15 Apr, 2020, 21:17 Giovanni Maruzzelli, <gmaruzz at gmail.com> wrote:

> at the bottom of the mail you find instruction on how to unsubscribe
>
>
>
>
> On Wed, Apr 15, 2020 at 5:38 PM Yan Zhang <yan at petatrading.com> wrote:
>
>> WHY I AM STILL RECEIVING THIS ENDLESS MAIL NOTICE?
>>
>> Please remove me. Thanks
>>
>>
>>
>> On Tuesday, April 14, 2020, 06:48:26 PM EDT, <mickael.bride at orange.com>
>> wrote:
>>
>>
>> Hello all,
>>
>>
>>
>> Sorry in advance if my email is not clear enough or relevant.
>>
>> My need is to have some kind of WebRTC gateway between a web application
>> and a call server managing SIP/RTP (the webapp is a softphone, audio only).
>>
>> This call server is not exposed on Internet, this is mainly why I need a
>> proxy/gateway.
>>
>>
>>
>> I am quite a newbie on WebRTC / SIP / RTP.
>>
>> From what I have seen on Internet some projects/products are doing that
>> kind of functionality (Asterisk, Doubango webrtc2sip, Janus, Audio Code …)
>> , but I am not able to focus on the simplest way to do it. Also some are
>> commercial products but I would prefer open source, installed on CentOS.
>>
>> FreeSwitch seems really great but I wonder if it is not too “big” or
>> complicated for my needs.
>>
>>
>>
>> Questions are:
>>
>> ·         Is FreeSwitch relevant to have a proxy/gateway between a
>> WebRTC web app and a call server not exposed on Internet?
>>
>> ·         Is it quite “simple” to install and configure freeSwitch to
>> have that functionality?
>>
>> ·         Is there any other simpler project to do this?
>>
>>
>>
>> Any advice would be really appreciate.
>>
>>
>>
>> Thank you,
>>
>>
>>
>> Mickaël
>>
>>
>>
>> _________________________________________________________________________________________________________________________
>>
>> Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc
>> pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler
>> a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration,
>> Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci.
>>
>> This message and its attachments may contain confidential or privileged information that may be protected by law;
>> they should not be distributed, used or copied without authorisation.
>> If you have received this email in error, please notify the sender and delete this message and its attachments.
>> As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified.
>> Thank you.
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200415/b92c0be2/attachment-0001.html>

From mrjoli021 at gmail.com  Wed Apr 15 16:09:10 2020
From: mrjoli021 at gmail.com (Joli Martinez)
Date: Wed, 15 Apr 2020 12:09:10 -0400
Subject: [Freeswitch-users] unable to post
Message-ID: <CANeteic6T5AZC6LBYpgnhH2a-x9vvux18-xU3BxG9CpZO_NzOw@mail.gmail.com>

Hello,

I am trying to figure out what the guidelines to post stuff is.  When I ask
a question recently my questions have not been posted.  Am I not
asking them in the correct way, or is there a limit to how many questions I
can post within a given time frame?

If that is the case could I get some feedback as to what I am doing wrong?

I am an experienced freeswitch user in certain in areas and I am sometimes
able to fix thing on my own, but when I am stuck I would like to be able to
post stuff online to get help from the community.

Thanks,
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200415/114655ac/attachment.html>

From gmaruzz at gmail.com  Wed Apr 15 17:06:12 2020
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 15 Apr 2020 19:06:12 +0200
Subject: [Freeswitch-users] FreeSwitch for WebRTC to SIP/RTP gateway?
In-Reply-To: <CAAoGpGSpjkG=geRYNfpSXui1ai=JwS3pMW6cE_Cg+vjYN0oucw@mail.gmail.com>
References: <926519743.991603.1586965101525.ref@mail.yahoo.com>
 <926519743.991603.1586965101525@mail.yahoo.com>
 <CALXCt0r2kqY9QFjzBmmFUg_tvV_C506-oKzbW9Nt6_3+SQuikw@mail.gmail.com>
 <CAAoGpGSpjkG=geRYNfpSXui1ai=JwS3pMW6cE_Cg+vjYN0oucw@mail.gmail.com>
Message-ID: <CALXCt0rSbroqw=BpbBq8Y3=A3_ytXduhiEat5nfGy=kpX5yfyA@mail.gmail.com>

Ciao Mitul, nice to read from you my friend!!!!!!

( I believe someone from List admins will do )



On Wed, Apr 15, 2020 at 6:47 PM Mitul Limbani <mitul at enterux.com> wrote:

> Someone please do him a favor by kicking out
>
> ML
>
> On Wed, 15 Apr, 2020, 21:17 Giovanni Maruzzelli, <gmaruzz at gmail.com>
> wrote:
>
>> at the bottom of the mail you find instruction on how to unsubscribe
>>
>>
>>
>>
>> On Wed, Apr 15, 2020 at 5:38 PM Yan Zhang <yan at petatrading.com> wrote:
>>
>>> WHY I AM STILL RECEIVING THIS ENDLESS MAIL NOTICE?
>>>
>>> Please remove me. Thanks
>>>
>>>
>>>
>>> On Tuesday, April 14, 2020, 06:48:26 PM EDT, <mickael.bride at orange.com>
>>> wrote:
>>>
>>>
>>> Hello all,
>>>
>>>
>>>
>>> Sorry in advance if my email is not clear enough or relevant.
>>>
>>> My need is to have some kind of WebRTC gateway between a web application
>>> and a call server managing SIP/RTP (the webapp is a softphone, audio only).
>>>
>>> This call server is not exposed on Internet, this is mainly why I need a
>>> proxy/gateway.
>>>
>>>
>>>
>>> I am quite a newbie on WebRTC / SIP / RTP.
>>>
>>> From what I have seen on Internet some projects/products are doing that
>>> kind of functionality (Asterisk, Doubango webrtc2sip, Janus, Audio Code …)
>>> , but I am not able to focus on the simplest way to do it. Also some are
>>> commercial products but I would prefer open source, installed on CentOS.
>>>
>>> FreeSwitch seems really great but I wonder if it is not too “big” or
>>> complicated for my needs.
>>>
>>>
>>>
>>> Questions are:
>>>
>>> ·         Is FreeSwitch relevant to have a proxy/gateway between a
>>> WebRTC web app and a call server not exposed on Internet?
>>>
>>> ·         Is it quite “simple” to install and configure freeSwitch to
>>> have that functionality?
>>>
>>> ·         Is there any other simpler project to do this?
>>>
>>>
>>>
>>> Any advice would be really appreciate.
>>>
>>>
>>>
>>> Thank you,
>>>
>>>
>>>
>>> Mickaël
>>>
>>>
>>>
>>> _________________________________________________________________________________________________________________________
>>>
>>> Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc
>>> pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler
>>> a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration,
>>> Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci.
>>>
>>> This message and its attachments may contain confidential or privileged information that may be protected by law;
>>> they should not be distributed, used or copied without authorisation.
>>> If you have received this email in error, please notify the sender and delete this message and its attachments.
>>> As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified.
>>> Thank you.
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> OpenTelecom.IT
>> cell: +39 347 266 56 18
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200415/ccd56bfb/attachment.html>

From david.villasmil.work at gmail.com  Wed Apr 15 17:22:44 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 15 Apr 2020 18:22:44 +0100
Subject: [Freeswitch-users] unable to post
In-Reply-To: <CANeteic6T5AZC6LBYpgnhH2a-x9vvux18-xU3BxG9CpZO_NzOw@mail.gmail.com>
References: <CANeteic6T5AZC6LBYpgnhH2a-x9vvux18-xU3BxG9CpZO_NzOw@mail.gmail.com>
Message-ID: <CAFGRPVqQK+RU5xLFDTA_HoojE4VEGy4=j9Z4cfnoqGY0=j6UQw@mail.gmail.com>

Hello, we are receiving your emails.
At least this one.

On Wed, 15 Apr 2020 at 17:48, Joli Martinez <mrjoli021 at gmail.com> wrote:

> Hello,
>
> I am trying to figure out what the guidelines to post stuff is.  When I
> ask a question recently my questions have not been posted.  Am I not
> asking them in the correct way, or is there a limit to how many questions I
> can post within a given time frame?
>
> If that is the case could I get some feedback as to what I am doing wrong?
>
> I am an experienced freeswitch user in certain in areas and I am sometimes
> able to fix thing on my own, but when I am stuck I would like to be able to
> post stuff online to get help from the community.
>
> Thanks,
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200415/a3847527/attachment-0001.html>

From mitul at enterux.com  Wed Apr 15 18:03:28 2020
From: mitul at enterux.com (Mitul Limbani)
Date: Wed, 15 Apr 2020 23:33:28 +0530
Subject: [Freeswitch-users] FreeSwitch for WebRTC to SIP/RTP gateway?
In-Reply-To: <CALXCt0rSbroqw=BpbBq8Y3=A3_ytXduhiEat5nfGy=kpX5yfyA@mail.gmail.com>
References: <926519743.991603.1586965101525.ref@mail.yahoo.com>
 <926519743.991603.1586965101525@mail.yahoo.com>
 <CALXCt0r2kqY9QFjzBmmFUg_tvV_C506-oKzbW9Nt6_3+SQuikw@mail.gmail.com>
 <CAAoGpGSpjkG=geRYNfpSXui1ai=JwS3pMW6cE_Cg+vjYN0oucw@mail.gmail.com>
 <CALXCt0rSbroqw=BpbBq8Y3=A3_ytXduhiEat5nfGy=kpX5yfyA@mail.gmail.com>
Message-ID: <CAAoGpGQJtNO6QY3J-Q14A12axk9kzB7K_omSgNR4BRCyNAPZCA@mail.gmail.com>

Awesome, thanks Giovanni ... Hope all is well with you at Italy !

Regards,
Mitul Limbani,
Business Head,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mitul at enterux.in
DID: +91-22-71967196
Cell: +91-9820332422


On Wed, Apr 15, 2020 at 10:37 PM Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> Ciao Mitul, nice to read from you my friend!!!!!!
>
> ( I believe someone from List admins will do )
>
>
>
> On Wed, Apr 15, 2020 at 6:47 PM Mitul Limbani <mitul at enterux.com> wrote:
>
>> Someone please do him a favor by kicking out
>>
>> ML
>>
>> On Wed, 15 Apr, 2020, 21:17 Giovanni Maruzzelli, <gmaruzz at gmail.com>
>> wrote:
>>
>>> at the bottom of the mail you find instruction on how to unsubscribe
>>>
>>>
>>>
>>>
>>> On Wed, Apr 15, 2020 at 5:38 PM Yan Zhang <yan at petatrading.com> wrote:
>>>
>>>> WHY I AM STILL RECEIVING THIS ENDLESS MAIL NOTICE?
>>>>
>>>> Please remove me. Thanks
>>>>
>>>>
>>>>
>>>> On Tuesday, April 14, 2020, 06:48:26 PM EDT, <mickael.bride at orange.com>
>>>> wrote:
>>>>
>>>>
>>>> Hello all,
>>>>
>>>>
>>>>
>>>> Sorry in advance if my email is not clear enough or relevant.
>>>>
>>>> My need is to have some kind of WebRTC gateway between a web
>>>> application and a call server managing SIP/RTP (the webapp is a softphone,
>>>> audio only).
>>>>
>>>> This call server is not exposed on Internet, this is mainly why I need
>>>> a proxy/gateway.
>>>>
>>>>
>>>>
>>>> I am quite a newbie on WebRTC / SIP / RTP.
>>>>
>>>> From what I have seen on Internet some projects/products are doing that
>>>> kind of functionality (Asterisk, Doubango webrtc2sip, Janus, Audio Code …)
>>>> , but I am not able to focus on the simplest way to do it. Also some are
>>>> commercial products but I would prefer open source, installed on CentOS.
>>>>
>>>> FreeSwitch seems really great but I wonder if it is not too “big” or
>>>> complicated for my needs.
>>>>
>>>>
>>>>
>>>> Questions are:
>>>>
>>>> ·         Is FreeSwitch relevant to have a proxy/gateway between a
>>>> WebRTC web app and a call server not exposed on Internet?
>>>>
>>>> ·         Is it quite “simple” to install and configure freeSwitch to
>>>> have that functionality?
>>>>
>>>> ·         Is there any other simpler project to do this?
>>>>
>>>>
>>>>
>>>> Any advice would be really appreciate.
>>>>
>>>>
>>>>
>>>> Thank you,
>>>>
>>>>
>>>>
>>>> Mickaël
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________________________________________________________
>>>>
>>>> Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc
>>>> pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler
>>>> a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration,
>>>> Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci.
>>>>
>>>> This message and its attachments may contain confidential or privileged information that may be protected by law;
>>>> they should not be distributed, used or copied without authorisation.
>>>> If you have received this email in error, please notify the sender and delete this message and its attachments.
>>>> As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified.
>>>> Thank you.
>>>>
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>>
>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> OpenTelecom.IT
>>> cell: +39 347 266 56 18
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200415/0998ddda/attachment.html>

From gmaruzz at gmail.com  Wed Apr 15 18:07:44 2020
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 15 Apr 2020 20:07:44 +0200
Subject: [Freeswitch-users] FreeSwitch for WebRTC to SIP/RTP gateway?
In-Reply-To: <CAAoGpGQJtNO6QY3J-Q14A12axk9kzB7K_omSgNR4BRCyNAPZCA@mail.gmail.com>
References: <926519743.991603.1586965101525.ref@mail.yahoo.com>
 <926519743.991603.1586965101525@mail.yahoo.com>
 <CALXCt0r2kqY9QFjzBmmFUg_tvV_C506-oKzbW9Nt6_3+SQuikw@mail.gmail.com>
 <CAAoGpGSpjkG=geRYNfpSXui1ai=JwS3pMW6cE_Cg+vjYN0oucw@mail.gmail.com>
 <CALXCt0rSbroqw=BpbBq8Y3=A3_ytXduhiEat5nfGy=kpX5yfyA@mail.gmail.com>
 <CAAoGpGQJtNO6QY3J-Q14A12axk9kzB7K_omSgNR4BRCyNAPZCA@mail.gmail.com>
Message-ID: <CALXCt0qCexRkEMGH1gkpVaoSx6j2ZrBLx45b5UqyWUO2Y5syng@mail.gmail.com>

Yep, as you, we're in lockdown, but in high spirit!

Big hugs!!

-giovanni



On Wed, Apr 15, 2020 at 8:04 PM Mitul Limbani <mitul at enterux.com> wrote:

> Awesome, thanks Giovanni ... Hope all is well with you at Italy !
>
> Regards,
> Mitul Limbani,
> Business Head,
> Enterux Solutions Pvt. Ltd.
> 110 Reena Complex, Opp. Nathani Steel,
> Vidyavihar (W), Mumbai - 400 086. India
> http://www.enterux.com/
> http://www.entvoice.com/
> email: mitul at enterux.in
> DID: +91-22-71967196
> Cell: +91-9820332422
>
>
> On Wed, Apr 15, 2020 at 10:37 PM Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> Ciao Mitul, nice to read from you my friend!!!!!!
>>
>> ( I believe someone from List admins will do )
>>
>>
>>
>> On Wed, Apr 15, 2020 at 6:47 PM Mitul Limbani <mitul at enterux.com> wrote:
>>
>>> Someone please do him a favor by kicking out
>>>
>>> ML
>>>
>>> On Wed, 15 Apr, 2020, 21:17 Giovanni Maruzzelli, <gmaruzz at gmail.com>
>>> wrote:
>>>
>>>> at the bottom of the mail you find instruction on how to unsubscribe
>>>>
>>>>
>>>>
>>>>
>>>> On Wed, Apr 15, 2020 at 5:38 PM Yan Zhang <yan at petatrading.com> wrote:
>>>>
>>>>> WHY I AM STILL RECEIVING THIS ENDLESS MAIL NOTICE?
>>>>>
>>>>> Please remove me. Thanks
>>>>>
>>>>>
>>>>>
>>>>> On Tuesday, April 14, 2020, 06:48:26 PM EDT, <mickael.bride at orange.com>
>>>>> wrote:
>>>>>
>>>>>
>>>>> Hello all,
>>>>>
>>>>>
>>>>>
>>>>> Sorry in advance if my email is not clear enough or relevant.
>>>>>
>>>>> My need is to have some kind of WebRTC gateway between a web
>>>>> application and a call server managing SIP/RTP (the webapp is a softphone,
>>>>> audio only).
>>>>>
>>>>> This call server is not exposed on Internet, this is mainly why I need
>>>>> a proxy/gateway.
>>>>>
>>>>>
>>>>>
>>>>> I am quite a newbie on WebRTC / SIP / RTP.
>>>>>
>>>>> From what I have seen on Internet some projects/products are doing
>>>>> that kind of functionality (Asterisk, Doubango webrtc2sip, Janus, Audio
>>>>> Code …) , but I am not able to focus on the simplest way to do it. Also
>>>>> some are commercial products but I would prefer open source, installed on
>>>>> CentOS.
>>>>>
>>>>> FreeSwitch seems really great but I wonder if it is not too “big” or
>>>>> complicated for my needs.
>>>>>
>>>>>
>>>>>
>>>>> Questions are:
>>>>>
>>>>> ·         Is FreeSwitch relevant to have a proxy/gateway between a
>>>>> WebRTC web app and a call server not exposed on Internet?
>>>>>
>>>>> ·         Is it quite “simple” to install and configure freeSwitch to
>>>>> have that functionality?
>>>>>
>>>>> ·         Is there any other simpler project to do this?
>>>>>
>>>>>
>>>>>
>>>>> Any advice would be really appreciate.
>>>>>
>>>>>
>>>>>
>>>>> Thank you,
>>>>>
>>>>>
>>>>>
>>>>> Mickaël
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________________________________________________________
>>>>>
>>>>> Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc
>>>>> pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler
>>>>> a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration,
>>>>> Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci.
>>>>>
>>>>> This message and its attachments may contain confidential or privileged information that may be protected by law;
>>>>> they should not be distributed, used or copied without authorisation.
>>>>> If you have received this email in error, please notify the sender and delete this message and its attachments.
>>>>> As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified.
>>>>> Thank you.
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>>
>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>> https://signalwire.com
>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>> services.
>>>>> Build your next product on our scalable cloud platform.
>>>>>
>>>>> Join our online community to chat in real time
>>>>> https://signalwire.community
>>>>>
>>>>> Professional FreeSWITCH Services
>>>>> sales at freeswitch.com
>>>>> https://freeswitch.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> https://freeswitch.com/oss
>>>>> https://freeswitch.org/confluence
>>>>> https://cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> https://freeswitch.com
>>>>>
>>>>
>>>>
>>>> --
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> OpenTelecom.IT
>>>> cell: +39 347 266 56 18
>>>>
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> OpenTelecom.IT
>> cell: +39 347 266 56 18
>>
>>

-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200415/18b4260b/attachment-0001.html>

From per at wgtwo.com  Thu Apr 16 14:02:57 2020
From: per at wgtwo.com (Per Modin)
Date: Thu, 16 Apr 2020 16:02:57 +0200
Subject: [Freeswitch-users] unable to post
In-Reply-To: <CANeteic6T5AZC6LBYpgnhH2a-x9vvux18-xU3BxG9CpZO_NzOw@mail.gmail.com>
References: <CANeteic6T5AZC6LBYpgnhH2a-x9vvux18-xU3BxG9CpZO_NzOw@mail.gmail.com>
Message-ID: <20200416140257.mvzzxeasbvyrewxu@nitrogen>

On 2020-04-15 12:09 CEST, Joli Martinez wrote:
> I am trying to figure out what the guidelines to post stuff is. When I 
> ask a question recently my questions have not been posted. Am I not 
> asking them in the correct way, or is there a limit to how many 
> questions I can post within a given time frame?

These are the ones I see, but not every thread have answers.

* 2020-04-15 18:09 Joli Martinez [Freeswitch-users] unable to post
* 2020-04-14 18:20 Joli Martinez [Freeswitch-users] Changing RTP ports mid call
* 2020-03-31 20:07 Joli Martinez [Freeswitch-users] customer sip trunk
* 2020-02-20 17:44 Joli Martinez [Freeswitch-users] ha-error messages
* 2020-02-17 21:37 Joli Martinez [Freeswitch-users] HA Cluster build

You can check the list online[1] if they have been received or not. In 
my experiecnce it can sometimes take a while for the messages to appear 
on the list, I don't know if I've been unlucky of if the mails are 
manually approved.

[1]: http://lists.freeswitch.org/pipermail/freeswitch-users/

> I am an experienced freeswitch user in certain in areas and I am 
> sometimes able to fix thing on my own, but when I am stuck I would 
> like to be able to post stuff online to get help from the community.

I'm in the same boat, and I agree. If it has been too long without 
solving it IMHO feel free to bump your thread. I don't know if there are 
any formal guidelines for this.

Best,
Per.
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 833 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200416/124816fb/attachment.sig>

From vma at vallimamod.org  Thu Apr 16 15:10:19 2020
From: vma at vallimamod.org (Valli A. Vallimamod)
Date: Thu, 16 Apr 2020 17:10:19 +0200
Subject: [Freeswitch-users] Changing RTP ports mid call
In-Reply-To: <CANeteid2sAAGfeFjFw3uq0GTLdjMooD1tou6gMJsu309GeWOEg@mail.gmail.com>
References: <CANeteid2sAAGfeFjFw3uq0GTLdjMooD1tou6gMJsu309GeWOEg@mail.gmail.com>
Message-ID: <F0E93737-21BF-446C-8BEA-7C3160CA26BC@vallimamod.org>

Hi,

In your case freeswitch starts sending the rtp to the advertised ip/port from SDP (124.123.104.212:10488) but detects that the rtp from the phone is coming from another port (9658) so it switches to this port as it the one actually opened by the nat router. You can disable this feature with the channel variable disable_rtp_auto_adjust=true.

Also the "BUG" in the log line is related to freeswitch media bug, not an actual issue.


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


> On 14 Apr 2020, at 18:20, Joli Martinez <mrjoli021 at gmail.com> wrote:
> 
> Hello,
> 
> We have a phone app that ties back to our freeswitch servers.  Recently the android phones are getting one-way audio.  It works on iPhones and desktop clients.  Based on the pcap's and fs_cli I see that the audio ports changed after the sdp negotiations.  Not sure why this is happening also see that the codec is set to "BUG Codec PCMU:0"
> 
> Any Suggestions?
> 
> 2020-04-14 15:56:07.661344 [DEBUG] switch_ivr_async.c:1500 No silence detection configured; assuming start of speech
> 2020-04-14 15:56:07.701439 [DEBUG] switch_core_io.c:448 Setting BUG Codec PCMU:0
> 2020-04-14 15:56:07.701439 [DEBUG] switch_rtp.c:1887 rtcp_stats_init: audio ssrc[1321291818] base_seq[30216]
> 2020-04-14 15:56:07.881347 [INFO] switch_rtp.c:7231 Auto Changing audio port from 124.123.104.212:10488 to 124.123.104.212:9658
> 
> Thanks,
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



From gmaruzz at gmail.com  Thu Apr 16 15:34:34 2020
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 16 Apr 2020 17:34:34 +0200
Subject: [Freeswitch-users] Announcing SaraPhone,
	SIP WebRTC Open Source business phone
Message-ID: <CALXCt0qaDgo0gSmkKSy0uzK+5v3tS6KRJgXN78u2qFWL6JH4MA@mail.gmail.com>

My fellow VoIPers,

I am pleased to announce the early availability of:

SaraPhone
------------------

SaraPhone is a bare bone SIP WebRTC voice phone, complete with most
features real companies want to use in real world: HotDesking, Redial,
BLFs, MWI, DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer,
Notifications, running on all Browsers both on Desktop and SmartPhone.

SaraPhone is fully integrated with FusionPBX, the full-featured domain
based multi-tenant PBX and voice switch for FreeSwitch.

Based on SIP.js, SaraPhone works with all WebRTC compliant SIP proxies,
gateways, and servers (Asterisk, OpenSIPS, Kamailio, etc).

Initial author is Giovanni Maruzzelli, and SaraPhone gets its name from
Giovanni's wife, Sara Hosseini.

In addition to providing all of the usual DeskPhone functionality,
SaraPhone got:

   - Desktop Notification for Incoming Calls
   - Live MWI update
   - Real Time BLFs status update
   - BLF click to call
   - Caller Name and Number Display
   - Call Error Cause Display
   - AutoAnswer
   - Network Disconnect Reload
   - Show and Set Caller-ID (incoming-outbound)


You an find it in GitHub ( https://github.com/gmaruzz/saraphone ).

Anyone interested can play with it :).

Have fun,
giovanni


-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200416/f58ce832/attachment.html>

From vishalmpai at gmail.com  Thu Apr 16 17:20:45 2020
From: vishalmpai at gmail.com (Vishal Pai)
Date: Thu, 16 Apr 2020 22:50:45 +0530
Subject: [Freeswitch-users] Announcing SaraPhone,
 SIP WebRTC Open Source business phone
In-Reply-To: <CALXCt0qaDgo0gSmkKSy0uzK+5v3tS6KRJgXN78u2qFWL6JH4MA@mail.gmail.com>
References: <CALXCt0qaDgo0gSmkKSy0uzK+5v3tS6KRJgXN78u2qFWL6JH4MA@mail.gmail.com>
Message-ID: <CACOmXtCWKT_o6GYXVb_Y1cYKjvA7kFZLfHEoTN6US1EcFM-ecQ@mail.gmail.com>

Great...

Will definitely try it out.

On Thu, Apr 16, 2020 at 9:29 PM Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> My fellow VoIPers,
>
> I am pleased to announce the early availability of:
>
> SaraPhone
> ------------------
>
> SaraPhone is a bare bone SIP WebRTC voice phone, complete with most
> features real companies want to use in real world: HotDesking, Redial,
> BLFs, MWI, DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer,
> Notifications, running on all Browsers both on Desktop and SmartPhone.
>
> SaraPhone is fully integrated with FusionPBX, the full-featured domain
> based multi-tenant PBX and voice switch for FreeSwitch.
>
> Based on SIP.js, SaraPhone works with all WebRTC compliant SIP proxies,
> gateways, and servers (Asterisk, OpenSIPS, Kamailio, etc).
>
> Initial author is Giovanni Maruzzelli, and SaraPhone gets its name from
> Giovanni's wife, Sara Hosseini.
>
> In addition to providing all of the usual DeskPhone functionality,
> SaraPhone got:
>
>    - Desktop Notification for Incoming Calls
>    - Live MWI update
>    - Real Time BLFs status update
>    - BLF click to call
>    - Caller Name and Number Display
>    - Call Error Cause Display
>    - AutoAnswer
>    - Network Disconnect Reload
>    - Show and Set Caller-ID (incoming-outbound)
>
>
> You an find it in GitHub ( https://github.com/gmaruzz/saraphone ).
>
> Anyone interested can play with it :).
>
> Have fun,
> giovanni
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200416/4b27c54a/attachment.html>

From vishalmpai at gmail.com  Thu Apr 16 17:26:32 2020
From: vishalmpai at gmail.com (Vishal Pai)
Date: Thu, 16 Apr 2020 22:56:32 +0530
Subject: [Freeswitch-users] Mod avmd
Message-ID: <CACOmXtAYmp3HLre1Ckot7764Z5u7ODPcdaFJgi6dS17mzrvE+A@mail.gmail.com>

Hello All

Hope you are well and safe.

I am being playing around mod_avmd. I am originating calls by originate
command both are External calls on DID's through gateway. Is it possible to
detect the avmd on both the legs A and B using that above module. Currently
only one leg A is being detected.

Thank You
Vishal P.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200416/4522c1a6/attachment-0001.html>

From david.villasmil.work at gmail.com  Thu Apr 16 17:47:08 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 16 Apr 2020 18:47:08 +0100
Subject: [Freeswitch-users] Announcing SaraPhone,
 SIP WebRTC Open Source business phone
In-Reply-To: <CACOmXtCWKT_o6GYXVb_Y1cYKjvA7kFZLfHEoTN6US1EcFM-ecQ@mail.gmail.com>
References: <CALXCt0qaDgo0gSmkKSy0uzK+5v3tS6KRJgXN78u2qFWL6JH4MA@mail.gmail.com>
 <CACOmXtCWKT_o6GYXVb_Y1cYKjvA7kFZLfHEoTN6US1EcFM-ecQ@mail.gmail.com>
Message-ID: <CAFGRPVrjLKbTadk3JXYQO-tRzwotc8URUm-O2gcioMHicE5KZw@mail.gmail.com>

👏🏻👏🏻👏🏻👏🏻Nice!!!

On Thu, 16 Apr 2020 at 18:36, Vishal Pai <vishalmpai at gmail.com> wrote:

> Great...
>
> Will definitely try it out.
>
> On Thu, Apr 16, 2020 at 9:29 PM Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> My fellow VoIPers,
>>
>> I am pleased to announce the early availability of:
>>
>> SaraPhone
>> ------------------
>>
>> SaraPhone is a bare bone SIP WebRTC voice phone, complete with most
>> features real companies want to use in real world: HotDesking, Redial,
>> BLFs, MWI, DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer,
>> Notifications, running on all Browsers both on Desktop and SmartPhone.
>>
>> SaraPhone is fully integrated with FusionPBX, the full-featured domain
>> based multi-tenant PBX and voice switch for FreeSwitch.
>>
>> Based on SIP.js, SaraPhone works with all WebRTC compliant SIP proxies,
>> gateways, and servers (Asterisk, OpenSIPS, Kamailio, etc).
>>
>> Initial author is Giovanni Maruzzelli, and SaraPhone gets its name from
>> Giovanni's wife, Sara Hosseini.
>>
>> In addition to providing all of the usual DeskPhone functionality,
>> SaraPhone got:
>>
>>    - Desktop Notification for Incoming Calls
>>    - Live MWI update
>>    - Real Time BLFs status update
>>    - BLF click to call
>>    - Caller Name and Number Display
>>    - Call Error Cause Display
>>    - AutoAnswer
>>    - Network Disconnect Reload
>>    - Show and Set Caller-ID (incoming-outbound)
>>
>>
>> You an find it in GitHub ( https://github.com/gmaruzz/saraphone ).
>>
>> Anyone interested can play with it :).
>>
>> Have fun,
>> giovanni
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> OpenTelecom.IT
>> cell: +39 347 266 56 18
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200416/62222379/attachment.html>

From mitul at enterux.com  Thu Apr 16 18:21:15 2020
From: mitul at enterux.com (Mitul Limbani)
Date: Thu, 16 Apr 2020 23:51:15 +0530
Subject: [Freeswitch-users] Announcing SaraPhone,
 SIP WebRTC Open Source business phone
In-Reply-To: <CALXCt0qaDgo0gSmkKSy0uzK+5v3tS6KRJgXN78u2qFWL6JH4MA@mail.gmail.com>
References: <CALXCt0qaDgo0gSmkKSy0uzK+5v3tS6KRJgXN78u2qFWL6JH4MA@mail.gmail.com>
Message-ID: <CAAoGpGSzph5NMAmDNzLgvQrG2_w7zLVi8UG7UXYnf6M2O_sPqQ@mail.gmail.com>

Great Giovanni, shall definitely try this in our environment :-)

Mitul Limbani

On Thu, 16 Apr, 2020, 21:05 Giovanni Maruzzelli, <gmaruzz at gmail.com> wrote:

> My fellow VoIPers,
>
> I am pleased to announce the early availability of:
>
> SaraPhone
> ------------------
>
> SaraPhone is a bare bone SIP WebRTC voice phone, complete with most
> features real companies want to use in real world: HotDesking, Redial,
> BLFs, MWI, DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer,
> Notifications, running on all Browsers both on Desktop and SmartPhone.
>
> SaraPhone is fully integrated with FusionPBX, the full-featured domain
> based multi-tenant PBX and voice switch for FreeSwitch.
>
> Based on SIP.js, SaraPhone works with all WebRTC compliant SIP proxies,
> gateways, and servers (Asterisk, OpenSIPS, Kamailio, etc).
>
> Initial author is Giovanni Maruzzelli, and SaraPhone gets its name from
> Giovanni's wife, Sara Hosseini.
>
> In addition to providing all of the usual DeskPhone functionality,
> SaraPhone got:
>
>    - Desktop Notification for Incoming Calls
>    - Live MWI update
>    - Real Time BLFs status update
>    - BLF click to call
>    - Caller Name and Number Display
>    - Call Error Cause Display
>    - AutoAnswer
>    - Network Disconnect Reload
>    - Show and Set Caller-ID (incoming-outbound)
>
>
> You an find it in GitHub ( https://github.com/gmaruzz/saraphone ).
>
> Anyone interested can play with it :).
>
> Have fun,
> giovanni
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200416/f2b509d7/attachment.html>

From roberto.dellapasqua at live.com  Thu Apr 16 15:55:42 2020
From: roberto.dellapasqua at live.com (Roberto Della Pasqua)
Date: Thu, 16 Apr 2020 15:55:42 +0000
Subject: [Freeswitch-users] verto and conference for webrtc audio mcu
Message-ID: <AM0PR0502MB365022599BF18F5F09E555CD99D80@AM0PR0502MB3650.eurprd05.prod.outlook.com>

Hello,

after a lot researching about the best viable webrtc audio mcu conference server, I have discovered freeswitch, and seems excellent.

Can somebody explain me how to:

  1.  Setup a mcu audio conference with webrtc clients, dynamically opening and closing rooms (no need of signaling, this will be managed outside)
  2.  How to compile opus codec for maximum speed, eg. FFIXED instead of float (negligible audio difference, but large scaling)
  3.  Lib_verto and lib_conference are needed, allright?
  4.  I like manage the freeswitch with lib_socket, does you have examples perhaps?

Thank you for this amazing software and for help with hints for best config webrtc audio only mcu.

Kind regards.

Roberto Della Pasqua
www.dellapasqua.com<http://www.dellapasqua.com>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200416/1c9f5f1e/attachment-0001.html>

From gmaruzz at gmail.com  Thu Apr 16 20:08:11 2020
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 16 Apr 2020 22:08:11 +0200
Subject: [Freeswitch-users] verto and conference for webrtc audio mcu
In-Reply-To: <AM0PR0502MB365022599BF18F5F09E555CD99D80@AM0PR0502MB3650.eurprd05.prod.outlook.com>
References: <AM0PR0502MB365022599BF18F5F09E555CD99D80@AM0PR0502MB3650.eurprd05.prod.outlook.com>
Message-ID: <CALXCt0pSx4p5c0GXV9S7cP2tx1xcXUiUwp8VyP3_t4q5QGNPBA@mail.gmail.com>

Hello Roberto,

you may want to read the FreeSWITCH books by PacktPub, and then go to
official documentation and reference site https://freeswitch.org/confluence/

Good luck!

-giovanni



On Thu, Apr 16, 2020 at 10:01 PM Roberto Della Pasqua <
roberto.dellapasqua at live.com> wrote:

> Hello,
>
>
>
> after a lot researching about the best viable webrtc audio mcu conference
> server, I have discovered freeswitch, and seems excellent.
>
>
>
> Can somebody explain me how to:
>
>    1. Setup a mcu audio conference with webrtc clients, dynamically
>    opening and closing rooms (no need of signaling, this will be managed
>    outside)
>    2. How to compile opus codec for maximum speed, eg. FFIXED instead of
>    float (negligible audio difference, but large scaling)
>    3. Lib_verto and lib_conference are needed, allright?
>    4. I like manage the freeswitch with lib_socket, does you have
>    examples perhaps?
>
>
>
> Thank you for this amazing software and for help with hints for best
> config webrtc audio only mcu.
>
>
>
> Kind regards.
>
>
>
> Roberto Della Pasqua
>
> www.dellapasqua.com
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200416/bbb301e4/attachment.html>

From cpservicespb at gmail.com  Fri Apr 17 15:16:14 2020
From: cpservicespb at gmail.com (CpServiceSPb .)
Date: Fri, 17 Apr 2020 18:16:14 +0300
Subject: [Freeswitch-users] Device / extension state (status) functionality
 at Freeswitch or get User (Peer) / Gateway (Trunk) state
In-Reply-To: <CAN4xyZOs2SytDRv0vEiAq45z67ssNOydjSpjPY7uMSKscJMdEg@mail.gmail.com>
References: <CAN4xyZOWkThEP0twjpG8yCxu2z=hhspJyZkJBGypKUYMtTk7YQ@mail.gmail.com>
 <CAN4xyZOs2SytDRv0vEiAq45z67ssNOydjSpjPY7uMSKscJMdEg@mail.gmail.com>
Message-ID: <CAN4xyZOEs49JHVKbM1R7PjC+JbQq5mR=reF_OA9qjZ5PD5edEQ@mail.gmail.com>

There is Freeswitch 1.8.5 built from sources worked on x64 Linux.

I need to check / get internal users / trunck gateways state BEFORE using
it to perform different actions in response of got conditions:
1. the called number is not exists in the directory at all;
2. the called number exist in the directory, but not registered at the
time, for example don' t launch softphone;
3. the called number is inuse;
4. the called number is busy;
5. the called number is not answered due timeout
6. the called number is answered.

Is there functionality looks like device and / or extension (user) state at
Freeswitch as for example <dialstatus> or <channelstarus> ?

How is the best way to get these different states for internal users, for
each one and /or for trunk gateways, per each also ?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200417/5797a048/attachment.html>

From piotr at dataandsignal.com  Fri Apr 17 23:23:23 2020
From: piotr at dataandsignal.com (Piotr Gregor)
Date: Sat, 18 Apr 2020 00:23:23 +0100
Subject: [Freeswitch-users] Mod avmd
In-Reply-To: <CACOmXtAYmp3HLre1Ckot7764Z5u7ODPcdaFJgi6dS17mzrvE+A@mail.gmail.com>
References: <CACOmXtAYmp3HLre1Ckot7764Z5u7ODPcdaFJgi6dS17mzrvE+A@mail.gmail.com>
Message-ID: <CAGKWT2wZOAjG5WcjW=GiED_=KME7ZRHvPNHin_joUkgKgcu7yQ@mail.gmail.com>

Hi Vishal,

To have avmd running on a call leg, let's call that channel, i.e. RTP sent
and/or recvd to/from a given RTP endpoint you need to start avmd on that
channel:
https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd#mod_avmd-APPinterface
You can select direction for that channel on which avmd should be running
by using inbound_channel/outbound_channel settings:
https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd#mod_avmd-set

for example:
    <action application="avmd_start"
data="inbound_channel=0,outbound_channel=1"
/>
will start avmd on RTP sent and received to a given endpoint.

all the best,




Piotr Gregor
Software Engineer

M: (+44) 07483 866 525     L: (+44) 01256 597 470     www: dataandsignal.com






On Thu, Apr 16, 2020 at 7:14 PM Vishal Pai <vishalmpai at gmail.com> wrote:

> Hello All
>
> Hope you are well and safe.
>
> I am being playing around mod_avmd. I am originating calls by originate
> command both are External calls on DID's through gateway. Is it possible to
> detect the avmd on both the legs A and B using that above module. Currently
> only one leg A is being detected.
>
> Thank You
> Vishal P.
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200418/9e18368c/attachment.html>

From piotr at dataandsignal.com  Mon Apr 20 19:58:12 2020
From: piotr at dataandsignal.com (Piotr Gregor)
Date: Mon, 20 Apr 2020 20:58:12 +0100
Subject: [Freeswitch-users] Mod avmd
In-Reply-To: <CAGKWT2wZOAjG5WcjW=GiED_=KME7ZRHvPNHin_joUkgKgcu7yQ@mail.gmail.com>
References: <CACOmXtAYmp3HLre1Ckot7764Z5u7ODPcdaFJgi6dS17mzrvE+A@mail.gmail.com>
 <CAGKWT2wZOAjG5WcjW=GiED_=KME7ZRHvPNHin_joUkgKgcu7yQ@mail.gmail.com>
Message-ID: <CAGKWT2z=_1G000qR96EtnmW7F+39z5kZCLep3eZ4BW0Pw04uZA@mail.gmail.com>

Hi Vishal,

Correction.
In previous email it should read:

*for example: *
*    <action application="avmd_start"
data="inbound_channel=1,outbound_channel=1" />*
*will start avmd on RTP sent and received to a given endpoint.*

cheers,



Piotr Gregor
Software Engineer

M: (+44) 07483 866 525     L: (+44) 01256 597 470     www: dataandsignal.com






On Sat, Apr 18, 2020 at 12:23 AM Piotr Gregor <piotr at dataandsignal.com>
wrote:

> Hi Vishal,
>
> To have avmd running on a call leg, let's call that channel, i.e. RTP sent
> and/or recvd to/from a given RTP endpoint you need to start avmd on that
> channel:
> https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd#mod_avmd-APPinterface
> You can select direction for that channel on which avmd should be running
> by using inbound_channel/outbound_channel settings:
> https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd#mod_avmd-set
>
> for example:
>     <action application="avmd_start" data="inbound_channel=0,outbound_channel=1"
> />
> will start avmd on RTP sent and received to a given endpoint.
>
> all the best,
>
>
>
>
> Piotr Gregor
> Software Engineer
>
> M: (+44) 07483 866 525     L: (+44) 01256 597 470     www:
> dataandsignal.com
>
>
>
>
>
>
> On Thu, Apr 16, 2020 at 7:14 PM Vishal Pai <vishalmpai at gmail.com> wrote:
>
>> Hello All
>>
>> Hope you are well and safe.
>>
>> I am being playing around mod_avmd. I am originating calls by originate
>> command both are External calls on DID's through gateway. Is it possible to
>> detect the avmd on both the legs A and B using that above module. Currently
>> only one leg A is being detected.
>>
>> Thank You
>> Vishal P.
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200420/04e34684/attachment.html>

From aidar.kamalov at gmail.com  Tue Apr 21 11:33:08 2020
From: aidar.kamalov at gmail.com (Aidar Kamalov)
Date: Tue, 21 Apr 2020 14:33:08 +0300
Subject: [Freeswitch-users] freeswitch video conference recording black
	screen
Message-ID: <CAPXdxKh3DuZq47-suVZYCe-QTooAEsE2iz-6V9iy_HLZ2oKtLA@mail.gmail.com>

Hello, I am trying to record video conference:
<param name="auto-record"
value="$${recordings_dir}/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.mp4"/>

But I am getting black screen with audio only in recording file. Although I
can record video on every single channels:
uuid_record UUID start /var/lib/freeswitch/recordings/UUID.mp4

-- 
Aydar A. Kamalov
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200421/9079e406/attachment.html>

From Gilles at Sauvaire.com  Sat Apr 18 06:18:35 2020
From: Gilles at Sauvaire.com (Gilles SAUVAIRE)
Date: Sat, 18 Apr 2020 08:18:35 +0200
Subject: [Freeswitch-users] Device / extension state (status)
	functionality at Freeswitch or get User (Peer) / Gateway
	(Trunk) state
In-Reply-To: <CAN4xyZOEs49JHVKbM1R7PjC+JbQq5mR=reF_OA9qjZ5PD5edEQ@mail.gmail.com>
References: <CAN4xyZOWkThEP0twjpG8yCxu2z=hhspJyZkJBGypKUYMtTk7YQ@mail.gmail.com>
 <CAN4xyZOs2SytDRv0vEiAq45z67ssNOydjSpjPY7uMSKscJMdEg@mail.gmail.com>
 <CAN4xyZOEs49JHVKbM1R7PjC+JbQq5mR=reF_OA9qjZ5PD5edEQ@mail.gmail.com>
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAAAyq5Nx/9DRDrZ3hLNfSk2jCgAAAEAAAAPM2fL121RhLjbAeLz+bRLoBAAAAAA==@Sauvaire.com>

Hello,

 

a track for you ...

I think that many of the answers to your questions are simply found in the CDRs. 

you can set and adjust all the fields you need ...

 

Gilles  

 

 

De : FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de CpServiceSPb .
Envoyé : vendredi 17 avril 2020 17:16
À : freeswitch-users at lists.freeswitch.org
Objet : [Freeswitch-users] Device / extension state (status) functionality at Freeswitch or get User (Peer) / Gateway (Trunk) state

 

There is Freeswitch 1.8.5 built from sources worked on x64 Linux.

 

I need to check / get internal users / trunck gateways state BEFORE using it to perform different actions in response of got conditions:
1. the called number is not exists in the directory at all;
2. the called number exist in the directory, but not registered at the time, for example don' t launch softphone;

3. the called number is inuse; 

4. the called number is busy;
5. the called number is not answered due timeout
6. the called number is answered.

 

Is there functionality looks like device and / or extension (user) state at Freeswitch as for example <dialstatus> or <channelstarus> ? 

 

How is the best way to get these different states for internal users, for each one and /or for trunk gateways, per each also ? 

 

 

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200418/fbaac4cb/attachment.html>

From lafuente_sonny at yahoo.com  Mon Apr 20 14:29:34 2020
From: lafuente_sonny at yahoo.com (Sonny Lafuente)
Date: Mon, 20 Apr 2020 22:29:34 +0800
Subject: [Freeswitch-users] Freeswitch drops calls after 32 seconds
In-Reply-To: <7dbddd3e-4eed-bde7-4b58-8bfa6cf7c96e@mikesdriveway.com>
References: <7dbddd3e-4eed-bde7-4b58-8bfa6cf7c96e@mikesdriveway.com>
Message-ID: <A286D849-E0F1-4A59-9858-55BB94A94E7D@yahoo.com>

Do you have outbound proxy configuration on your setup? I have a same issue with this 32 sec call dropped. I manage to fix it by doing the outbound configuation. Btw, my setup is I have sip server behind NAT. 

Leg A <-> Local SIP server <->NAT <-> Cloud SIP Server <-> Leg B

My local calls works fine. I configure the internal and external profile the ip of my local sip server 192.168.x.x

Thanks,

> On Apr 7, 2020, at 10:15 AM, MikeKulls <freeswitch at mikesdriveway.com> wrote:
> 
> I've installed freeswitch on a Raspberry PI 3 with pretty much bog stock config. The only change I've done is to delete the ipv6 profiles and changed the default password. I've connected a few different clients, 3 soft phones and 2 ATAs. When making calls with some of the clients the calls are dropped after 32 seconds. For other clients the calls will proceed for half an hour no issues. I have absolutely no complexity here, it's all internal phones on the local /24 lan. There are no firewalls, traffic only goes via a switch. I have no external SIP gateway, none of the clients are behind a NAT. I've done extensive googling and everything I find talks about NAT issues and VPNs, multiple subnets etc. I'm not running any of that. Packet capture and more details below.
> 
> 
> Some of the things I've tried
> 
> - complete OS and freeswitch reinstall on the PI
> 
> - disable IPv6 on the PI (made a big improvement, now the remote phone rings instantly instead of long delay)
> 
> - full OS update before installing freeswitch
> 
> - tried clients with both UDP and TCP
> 
> - install freeswitch on CentOS 7.
> 
> - Disable SIP ALG on router (traffic not going through router anyway)
> 
> - Tried all devices on a different network switch
> 
> - Tried client options like "use rport" etc
> 
> 
> 
> 
> What works:
> 
> - Cisco SPA3102 ATA
> 
> - Voiper latest version on Windows
> 
> 
> What doesn't work
> 
> - Latest Voiper on android
> 
> - Older version of voiper on windows
> 
> - Billion 7404VGP
> 
> - Android version of Voip by antisip
> 
> 
> 
> TShark capture of a call dropping. IPs are:
> 
> 192.168.1.245 Freeswitch
> 
> 192.168.1.225 Windows PC with older copy of voiper
> 
> 192.168.1.202 Cisco ATA
> 
> 
>     1 0.000000000 192.168.1.225 â 192.168.1.245 SIP/SDP 1038 Request: INVITE sip:1004 at 192.168.1.245;transport=UDP |
>     2 0.005573031 192.168.1.245 â 192.168.1.225 SIP 917 Status: 407 Proxy Authentication Required |
>     3 0.011167520 192.168.1.225 â 192.168.1.245 SIP 408 Request: ACK sip:1004 at 192.168.1.245;transport=UDP |
>     4 0.016217586 192.168.1.225 â 192.168.1.245 SIP/SDP 1314 Request: INVITE sip:1004 at 192.168.1.245;transport=UDP |
>     5 0.037157271 192.168.1.245 â 192.168.1.225 SIP 413 Status: 100 Trying |
>     6 0.094830064 192.168.1.245 â 192.168.1.202 SIP/SDP 1283 Request: INVITE sip:1004 at 192.168.1.202:5083;transport=tcp |
>     7 0.115792718 192.168.1.202 â 192.168.1.245 SIP 365 Status: 100 Trying |
>     8 0.115932040 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083 [ACK] Seq=1230 Ack=312 Win=63784 Len=0
>     9 0.124784941 192.168.1.202 â 192.168.1.245 SIP 526 Status: 180 Ringing |
>    10 0.124863118 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083 [ACK] Seq=1230 Ack=784 Win=63784 Len=0
>    11 0.151217241 192.168.1.245 â 192.168.1.225 SIP/SDP 1205 Status: 183 Session Progress |
>    12 2.263254019 192.168.1.202 â 192.168.1.245 SIP/SDP 901 Status: 200 OK |
>    13 2.263406934 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083 [ACK] Seq=1230 Ack=1631 Win=63784 Len=0
>    14 2.272446033 192.168.1.245 â 192.168.1.202 SIP 471 Request: ACK sip:1004 at 192.168.1.202:5083;transport=tcp |
>    15 2.284211777 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK |
>    16 2.469876726 192.168.1.202 â 192.168.1.245 TCP 60 5083 â 5060 [ACK] Seq=1631 Ack=1647 Win=16000 Len=0
>    17 2.784653094 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK |
>    18 3.785916037 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK |
>    19 5.787085997 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK |
>    20 8.903231371 192.168.1.245 â 192.168.1.225 TCP 54 5060 â 56460 [ACK] Seq=1 Ack=1 Win=501 Len=0
>    21 8.903781836 192.168.1.225 â 192.168.1.245 TCP 60 [TCP ACKed unseen segment] 56460 â 5060 [ACK] Seq=1 Ack=2 Win=8212 Len=0
>    22 9.788128841 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK |
>    23 13.788284085 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK |
>    24 17.788442453 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK |
>    25 20.361180947 192.168.1.225 â 192.168.1.245 UDP 60 5060 â 5060 Len=4
>    26 21.788537538 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK |
>    27 25.788690954 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK |
>    28 29.789847278 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK |
>    29 32.583226442 192.168.1.245 â 192.168.1.202 TCP 54 [TCP Keep-Alive] 5060 â 5083 [ACK] Seq=1646 Ack=1631 Win=63784 Len=0
>    30 32.584076227 192.168.1.202 â 192.168.1.245 TCP 60 [TCP Keep-Alive ACK] 5083 â 5060 [ACK] Seq=1631 Ack=1647 Win=16000 Len=0
>    31 33.790019961 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK |
>    32 34.284637050 192.168.1.245 â 192.168.1.225 SIP 704 Request: BYE sip:1002 at 192.168.1.225:5060;transport=UDP |
>    33 34.327728390 192.168.1.245 â 192.168.1.202 SIP 672 Request: BYE sip:1004 at 192.168.1.202:5083;transport=tcp |
>    34 34.344437951 192.168.1.202 â 192.168.1.245 SIP 381 Status: 200 OK |
>    35 34.393228415 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083 [ACK] Seq=2265 Ack=1958 Win=63784 Len=0
>    36 34.422931003 192.168.1.225 â 192.168.1.245 SIP 466 Status: 200 OK |
>    37 38.352414375 192.168.1.225 â 192.168.1.245 TCP 60 [TCP ACKed unseen segment] 56460 â 5060 [PSH, ACK] Seq=1 Ack=2 Win=8212 Len=4
>    38 38.352610259 192.168.1.245 â 192.168.1.225 TCP 56 [TCP Previous segment not captured] 5060 â 56460 [PSH, ACK] Seq=2 Ack=5 Win=501 Len=2
>    39 38.393526512 192.168.1.225 â 192.168.1.245 TCP 60 [TCP ACKed unseen segment] 56460 â 5060 [ACK] Seq=5 Ack=4 Win=8212 Len=0
> 
> -- 
> This email has been checked for viruses by AVG.
> https://www.avg.com
> 
> 
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



From petedao at gmail.com  Sat Apr 18 07:36:50 2020
From: petedao at gmail.com (Pete Kay)
Date: Sat, 18 Apr 2020 00:36:50 -0700
Subject: [Freeswitch-users] send_dtmf only sends last 2 digits
Message-ID: <CAPxXe8WPk5GOQp0u9icBCcdVTqVTn+4uZuOrVX=0nu-HyGE_2Q@mail.gmail.com>

Hi
I used the following command to send DTMF to send to a device:

send_dtmf WWWWWW98717288077WW597177



The device only can pick up "77".


Could someone please tell me why and how to resolve it ?


Thanks,

Pete
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200418/08b39df1/attachment-0001.html>

From seanmhansen7 at gmail.com  Wed Apr 22 00:50:36 2020
From: seanmhansen7 at gmail.com (Sean Hansen)
Date: Tue, 21 Apr 2020 17:50:36 -0700
Subject: [Freeswitch-users] sofia reject MESSAGE method
Message-ID: <CAF=KPsNJDr0CjhVpRs0qKZnsigGKoTX95Dw5=n_tc3wy2gFvZg@mail.gmail.com>

Hello,

Does anyone know of a way to reject chat/messaging for sofia? Specifically
the MESSAGE method?

We use FreeSWITCH purely as an SBC/media proxy for voice only, so therefore
have no need for directory/chatplan/etc. We have noticed with our current
configuration, when our PBX passes a MESSAGE (usually generated on
accident...) to freeswitch it sends the MESSAGE right back to the PBX -
which creates a loop and jams up the proxies.

The log is complaining as such:
freeswitch.log.2020-04-21-11-57-04.1:2020-04-21 10:50:43.195480 [DEBUG]
sofia_presence.c:225 Can't find registered user XXXXXXXXXXX at domain.com
freeswitch.log.2020-04-21-11-57-04.1:2020-04-21 10:50:43.195480 [DEBUG]
sofia_presence.c:225 Can't find registered user XXXXXXXXXXX at domain.com
freeswitch.log.2020-04-21-11-57-04.1:2020-04-21 10:50:43.215485 [DEBUG]
sofia_presence.c:225 Can't find registered user XXXXXXXXXXX at domain.com
freeswitch.log.2020-04-21-11-57-04.1:2020-04-21 10:50:43.215485 [DEBUG]
sofia_presence.c:225 Can't find registered user XXXXXXXXXXX at domain.com

where XXXXXXXXXXX is the destination number for the message.

We do not load mod_sms, and all of the chatplans are disabled.

Thanks.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200421/12107789/attachment.html>

From sobomax at sippysoft.com  Tue Apr 21 06:57:09 2020
From: sobomax at sippysoft.com (Maxim Sobolev)
Date: Mon, 20 Apr 2020 23:57:09 -0700
Subject: [Freeswitch-users] Join us for SIP Chronicles Live #1,
	featuring Giovanni Maruzzelli
Message-ID: <CAH7qZfsT1N6omRZ8K9sC8VXQOhh57Z_nuFx3Tvxm9SkZpUprXQ@mail.gmail.com>

Dear Real-Time Friends and Colleagues!


As many of you we have been totally devastated that we will have no chance
to see you in the next few months to come. :-/ Some people in the community
believe it might be years. I don’t necessarily agree with that opinion
myself.

Over the course of the last few years our team had a great time extending
live coverage for some of those events that have been affected, got some
experience and equipment. Instead of just waiting for the virus to clear,
we decided to organize a series of bi-weekly live casts with some of the
speakers that we have hoped to see at those events presenting their latest
developments live and then answering questions from the audience.

So without further ado, let me introduce our first guest Giovanni
Maruzzelli, who is going to introduce his newest project SaraPhone (
https://github.com/gmaruzz/saraphone).

Join us this Saturday, April 25th 4:30pm UTC and get a chance to ask
Giovanni a question about his project live:

https://youtu.be/mF9elIcVGE8

Or if you miss that opportunity, you can always watch the recording later
on Sippy Labs channel on YouTube and email Giovanni your question at <
gmaruzz at gmail.com>.

SaraPhone is a bare bone SIP WebRTC phone, complete with most features real
companies want to use in real world: HotDesking, Redial, BLFs, MWI, DND,
PhoneBook, Hold, Transfer, Mute, Attended Transfer, Notifications, running
on all Browsers both on Desktop and SmartPhone.

SaraPhone is fully integrated with FusionPBX, the full-featured domain
based multi-tenant PBX and voice switch for FreeSwitch.

Based on SIP.js, SaraPhone works with all WebRTC compliant SIP proxies,
gateways, and servers (Asterisk, OpenSIPS, Kamailio, Janus, etc).

Initial author is Giovanni Maruzzelli, and SaraPhone gets its name from
Giovanni's wife, Sara Hosseini.

In addition to providing all of the usual DeskPhone functionality,
SaraPhone got:

   -

   Desktop Notification for Incoming Calls
   -

   Live MWI update
   -

   Real Time BLFs status update
   -

   BLF click to call
   -

   Caller Name and Number Display
   -

   Call Error Cause Display
   -

   AutoAnswer
   -

   Network Disconnect Reload
   -

   Show and Set Caller-ID (incoming-outbound)

Stay healthy, optimistic and productive! Also share, like and subscribe.
See you soon!!!

Regards,

Max
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200420/8fe447cd/attachment.html>

From loidang at hoiio.com  Wed Apr 22 04:00:19 2020
From: loidang at hoiio.com (Loi Dang)
Date: Wed, 22 Apr 2020 11:00:19 +0700
Subject: [Freeswitch-users] sofia reject MESSAGE method
In-Reply-To: <CAF=KPsNJDr0CjhVpRs0qKZnsigGKoTX95Dw5=n_tc3wy2gFvZg@mail.gmail.com>
References: <CAF=KPsNJDr0CjhVpRs0qKZnsigGKoTX95Dw5=n_tc3wy2gFvZg@mail.gmail.com>
Message-ID: <CADMnEC-B64+xPjK9hrmoTcPP_2ZTb-N-49zMoqfUMg4GTUWs9A@mail.gmail.com>

Hi, try setting this var `enable-chat` to false in your sofia sip profile.

rgds,
Loi Dang

On Wed, Apr 22, 2020 at 10:32 AM Sean Hansen <seanmhansen7 at gmail.com> wrote:

> Hello,
>
> Does anyone know of a way to reject chat/messaging for sofia? Specifically
> the MESSAGE method?
>
> We use FreeSWITCH purely as an SBC/media proxy for voice only, so
> therefore have no need for directory/chatplan/etc. We have noticed with our
> current configuration, when our PBX passes a MESSAGE (usually generated on
> accident...) to freeswitch it sends the MESSAGE right back to the PBX -
> which creates a loop and jams up the proxies.
>
> The log is complaining as such:
> freeswitch.log.2020-04-21-11-57-04.1:2020-04-21 10:50:43.195480 [DEBUG]
> sofia_presence.c:225 Can't find registered user XXXXXXXXXXX at domain.com
> freeswitch.log.2020-04-21-11-57-04.1:2020-04-21 10:50:43.195480 [DEBUG]
> sofia_presence.c:225 Can't find registered user XXXXXXXXXXX at domain.com
> freeswitch.log.2020-04-21-11-57-04.1:2020-04-21 10:50:43.215485 [DEBUG]
> sofia_presence.c:225 Can't find registered user XXXXXXXXXXX at domain.com
> freeswitch.log.2020-04-21-11-57-04.1:2020-04-21 10:50:43.215485 [DEBUG]
> sofia_presence.c:225 Can't find registered user XXXXXXXXXXX at domain.com
>
> where XXXXXXXXXXX is the destination number for the message.
>
> We do not load mod_sms, and all of the chatplans are disabled.
>
> Thanks.
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200422/31d6f7c5/attachment-0001.html>

From covici at ccs.covici.com  Wed Apr 22 07:51:47 2020
From: covici at ccs.covici.com (John Covici)
Date: Wed, 22 Apr 2020 03:51:47 -0400
Subject: [Freeswitch-users] send_dtmf only sends last 2 digits
In-Reply-To: <CAPxXe8WPk5GOQp0u9icBCcdVTqVTn+4uZuOrVX=0nu-HyGE_2Q@mail.gmail.com>
References: <CAPxXe8WPk5GOQp0u9icBCcdVTqVTn+4uZuOrVX=0nu-HyGE_2Q@mail.gmail.com>
Message-ID: <m3o8rkt0to.wl-covici@ccs.covici.com>

Try putting a w before each digit, may need more than one depending to
whom you are sending.

On Sat, 18 Apr 2020 03:36:50 -0400,
Pete Kay wrote:
> 
> [1  <multipart/alternative (7bit)>]
> [1.1  <text/plain; UTF-8 (7bit)>]
> Hi
> I used the following command to send DTMF to send to a device:
> 
> send_dtmf WWWWWW98717288077WW597177
> 
> 
> 
> The device only can pick up "77".
> 
> 
> Could someone please tell me why and how to resolve it ?
> 
> 
> Thanks,
> 
> Pete
> [1.2  <text/html; UTF-8 (quoted-printable)>]
> [2  <text/plain; utf-8 (base64)>]
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici wb2una
         covici at ccs.covici.com


From david.villasmil.work at gmail.com  Thu Apr 23 10:52:04 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 23 Apr 2020 11:52:04 +0100
Subject: [Freeswitch-users] Add body to INFO reply
Message-ID: <CAFGRPVpjjqe8oFve5uQBPM9bEY8+41GmiEicsNq9UNoJG0fRJw@mail.gmail.com>

Hello all,

Is it possible to add an xml body (or whatever) to an incoming INFO request?

Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200423/f147ec3d/attachment.html>

From rw at panorgan.ch  Thu Apr 23 11:09:45 2020
From: rw at panorgan.ch (=?UTF-8?Q?Ren=c3=a9_Weiss?=)
Date: Thu, 23 Apr 2020 13:09:45 +0200
Subject: [Freeswitch-users] Wrong number of tries with originate_retries?
Message-ID: <292c6c28-895d-94bb-a37d-53293a5f5534@panorgan.ch>

Hi,

The documentation for originate_retries says "Number of retries before
giving up on originating a call (default is 0)."

Because of this I would expect that:

- originate_retries=0 = 1 call
- originate_retries=1 = 2 calls (1 try and 1 retry)
- originate_retries=2 = 3 calls (1 try and 2 retries)
  -originate_retries=3 = 4 calls (1 try and 3 retries)

But in my testing I get:

- originate_retries=0 = 1 call
- originate_retries=1 = 1 call
- originate_retries=2 = 2 calls
- originate_retries=3 = 3 calls

My originate command is:

originate{originate_timeout=5,originate_retries=<X>,originate_retry_sleep_ms=5000}user/662 &park()

And my FreeSWITCH version is:

FreeSWITCH Version 1.10.2-release-14-f7bdd3845a~64bit (-release-14-f7bdd3845a 64bit)

Am I doing something wrong here, do I understand the documentation wrong,
or is the problem in freeswitch / the freeswitch documentation?

Thanks.



From david.villasmil.work at gmail.com  Thu Apr 23 13:43:46 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 23 Apr 2020 14:43:46 +0100
Subject: [Freeswitch-users] Add body to INFO reply
In-Reply-To: <CAFGRPVpjjqe8oFve5uQBPM9bEY8+41GmiEicsNq9UNoJG0fRJw@mail.gmail.com>
References: <CAFGRPVpjjqe8oFve5uQBPM9bEY8+41GmiEicsNq9UNoJG0fRJw@mail.gmail.com>
Message-ID: <CAFGRPVqE8yHNX5L3-13ei_Josy0FbX4wOkj8a4EMKtVdKW2Jcw@mail.gmail.com>

Anyone? :D

On Thu, 23 Apr 2020 at 11:52, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello all,
>
> Is it possible to add an xml body (or whatever) to an incoming INFO
> request?
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
>
-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200423/b066cced/attachment.html>

From davidswalkabout at gmail.com  Thu Apr 23 17:47:32 2020
From: davidswalkabout at gmail.com (David P)
Date: Thu, 23 Apr 2020 10:47:32 -0700
Subject: [Freeswitch-users] Auth rejected after upgrade to 1.10
Message-ID: <CAE23trthdMdi177A+vdhbedHNfSLiM7m=GVGpZewjPFVsAvbHw@mail.gmail.com>

A few questions I've asked before...

1) We're evaluating an upgrade to 1.10 from 1.8, and we found that verto
can wss login but calls trigger "code": -32000 authentication required.
What can we do to fix this?

2) We sometimes get MEDIA_TIMEOUT from our automated peer in a 1-on-1
conference. Can we use execute_on_media_timeout to attempt a new conference
with the peer?

3) Can FS be made to provide TCP ice candidates in its answer sdp? (That
is, does it support ICE TCP?) For users behind firewalls that block many
UDP ports, this would allow avoiding added cost of TURN TCP.

Cheers,
David
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200423/acf640b9/attachment.html>

From cpservicespb at gmail.com  Wed Apr 22 11:27:25 2020
From: cpservicespb at gmail.com (CpServiceSPb .)
Date: Wed, 22 Apr 2020 14:27:25 +0300
Subject: [Freeswitch-users] Device / extension state (status)
 functionality at Freeswitch or get User (Peer) / Gateway (Trunk) state
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAAAyq5Nx/9DRDrZ3hLNfSk2jCgAAAEAAAAPM2fL121RhLjbAeLz+bRLoBAAAAAA==@Sauvaire.com>
References: <CAN4xyZOWkThEP0twjpG8yCxu2z=hhspJyZkJBGypKUYMtTk7YQ@mail.gmail.com>
 <CAN4xyZOs2SytDRv0vEiAq45z67ssNOydjSpjPY7uMSKscJMdEg@mail.gmail.com>
 <CAN4xyZOEs49JHVKbM1R7PjC+JbQq5mR=reF_OA9qjZ5PD5edEQ@mail.gmail.com>
 <!&!AAAAAAAAAAAYAAAAAAAAAAyq5Nx/9DRDrZ3hLNfSk2jCgAAAEAAAAPM2fL121RhLjbAeLz+bRLoBAAAAAA==@Sauvaire.com>
Message-ID: <CAN4xyZPq_aHW1efRzpjcFdV-XV6QrOEZ3DqA9Yu5UgjJkWSBXg@mail.gmail.com>

May you explain more detailed what did you imply for the case ?

For example there are some internal peers = 100, 101 and some external
trunks = VoIPProv1, VoIPProv2.
User with 100 tries to call user with 101.
And just before calling how is to get 101 peer status/state using CDR to
implemen logic - for example, to redirect call to peer 101 cell number in
case of 101 busy answer or to speak peer 100 message that peer 101 is busy ?
The same is for external trunks.
How is to do si by CDR as for out as for in coming calling as for internal
peers as for external gateways ?

ср, 22 апр. 2020 г. в 05:53, Gilles SAUVAIRE <Gilles at sauvaire.com>:

> Hello,
>
>
>
> a track for you ...
>
> I think that many of the answers to your questions are simply found in the
> CDRs.
>
> you can set and adjust all the fields you need ...
>
>
>
> Gilles
>
>
>
>
>
> *De :* FreeSWITCH-users [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *De la part de*
> CpServiceSPb .
> *Envoyé :* vendredi 17 avril 2020 17:16
> *À :* freeswitch-users at lists.freeswitch.org
> *Objet :* [Freeswitch-users] Device / extension state (status)
> functionality at Freeswitch or get User (Peer) / Gateway (Trunk) state
>
>
>
> There is Freeswitch 1.8.5 built from sources worked on x64 Linux.
>
>
>
> I need to check / get internal users / trunck gateways state BEFORE using
> it to perform different actions in response of got conditions:
> 1. the called number is not exists in the directory at all;
> 2. the called number exist in the directory, but not registered at the
> time, for example don' t launch softphone;
>
> 3. the called number is inuse;
>
> 4. the called number is busy;
> 5. the called number is not answered due timeout
> 6. the called number is answered.
>
>
>
> Is there functionality looks like device and / or extension (user) state
> at Freeswitch as for example <dialstatus> or <channelstarus> ?
>
>
>
> How is the best way to get these different states for internal users, for
> each one and /or for trunk gateways, per each also ?
>
>
>
>
>
>
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200422/c683e726/attachment.html>

From vma at vallimamod.org  Fri Apr 24 10:35:10 2020
From: vma at vallimamod.org (Valli A. Vallimamod)
Date: Fri, 24 Apr 2020 12:35:10 +0200
Subject: [Freeswitch-users] Wrong number of tries with originate_retries?
In-Reply-To: <292c6c28-895d-94bb-a37d-53293a5f5534@panorgan.ch>
References: <292c6c28-895d-94bb-a37d-53293a5f5534@panorgan.ch>
Message-ID: <245C8FF5-824D-4807-817F-7657F5C70B86@vallimamod.org>

Hi,

In the source code (switch_ivr_originate() in switch_ivr_originate.c from latest master branch), valid values for originate_retries are between 1 and 100 with default to 1.

And the originate loop looks like this:

  for (try = 0; try < originate_retries; try++) {
      if (try > 0)
          wait(originate_retry_min_period_ms);
      dial();
  }

So its compliant with what you observe. The name of the variable may be misleading and the documentation looks incorrect.



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


> On 23 Apr 2020, at 13:09, René Weiss <rw at panorgan.ch> wrote:
> 
> Hi,
> 
> The documentation for originate_retries says "Number of retries before
> giving up on originating a call (default is 0)."
> 
> Because of this I would expect that:
> 
> - originate_retries=0 = 1 call
> - originate_retries=1 = 2 calls (1 try and 1 retry)
> - originate_retries=2 = 3 calls (1 try and 2 retries)
> -originate_retries=3 = 4 calls (1 try and 3 retries)
> 
> But in my testing I get:
> 
> - originate_retries=0 = 1 call
> - originate_retries=1 = 1 call
> - originate_retries=2 = 2 calls
> - originate_retries=3 = 3 calls
> 
> My originate command is:
> 
> originate{originate_timeout=5,originate_retries=<X>,originate_retry_sleep_ms=5000}user/662 &park()
> 
> And my FreeSWITCH version is:
> 
> FreeSWITCH Version 1.10.2-release-14-f7bdd3845a~64bit (-release-14-f7bdd3845a 64bit)
> 
> Am I doing something wrong here, do I understand the documentation wrong,
> or is the problem in freeswitch / the freeswitch documentation?
> 
> Thanks.
> 
> 
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



From Jazmin.Marino at on24.com  Fri Apr 24 19:54:45 2020
From: Jazmin.Marino at on24.com (Jazmin Marina Florez Marino)
Date: Fri, 24 Apr 2020 19:54:45 +0000
Subject: [Freeswitch-users] ADD 2 ICE CANDIDATES  mod_verto
Message-ID: <EAA0402198B6144FA20A2B37EB4E1A656E01E3E2@R-ESXMBX01.on24.com>

Hello all,

I am working with mod_verto in a scenario where I need to send public and private IP addresses as ICE candidates.
When I modify the ext-rtp-ip parameter, it only sends one option as typ host ice candidate. However, for my implementation I need to send both.

What I need is the combination of those ice candidates, something like this:
a=candidate:8228103366 1 udp 659136 190.84.118.88 31028 typ host generation 0
a=candidate:3496438569 1 udp 659136 192.168.1.170 24210 typ host generation 0

Is it possible to add both ice candidates in the same SDP from configuration files?


SDP with local IP
2020-04-24 14:17:23.715620 [DEBUG] mod_verto.c:2502 Local SDP verto.rtc/5715976:
v=0
o=FreeSWITCH 1587724815 1587724816 IN IP4 190.84.118.88
s=FreeSWITCH
c=IN IP4 190.84.118.88
t=0 0
a=msid-semantic: WMS 8gumHFWDC0sD12iPvVhSeEhUfLdXQBVO
m=audio 31028 UDP/TLS/RTP/SAVPF 111 110
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1; minptime=10; stereo=1; sprop-stereo=1
a=rtpmap:110 telephone-event/48000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=fingerprint:sha-256 FD:DF:48:02:C3:F8:D9:57:F8:C8:97:CD:3A:AB:6A:A6:97:4A:BA:A5:48:5A:3A:2D:0F:18:14:EC:D5:A9:21:86
a=setup:active
a=rtcp-mux
a=rtcp:31028 IN IP4 190.84.118.88
a=ice-ufrag:BiYFM6pSItFoUq2t
a=ice-pwd:wTz7x9b00ImLonRPvGl8CBRb
a=candidate:8228103366 1 udp 659136 190.84.118.88 31028 typ host generation 0
a=end-of-candidates
a=ssrc:2930644107 cname:If1gLNJdD1HvUPbg
a=ssrc:2930644107 msid:8gumHFWDC0sD12iPvVhSeEhUfLdXQBVO a0
a=ssrc:2930644107 mslabel:8gumHFWDC0sD12iPvVhSeEhUfLdXQBVO
a=ssrc:2930644107 label:8gumHFWDC0sD12iPvVhSeEhUfLdXQBVOa0
m=video 0 UDP/TLS/RTP/SAVPF 19

SDP with Public IP
<param name="ext-rtp-ip" value="190.84.118.88"/>
2020-04-24 14:13:49.395900 [DEBUG] mod_verto.c:2502 Local SDP verto.rtc/5715976:
v=0
o=FreeSWITCH 1587731419 1587731420 IN IP4 192.168.1.170
s=FreeSWITCH
c=IN IP4 192.168.1.170
t=0 0
a=msid-semantic: WMS t9mERteVPrwT3aZbexPlp2U6Cspg78dq
m=audio 24210 UDP/TLS/RTP/SAVPF 111 110
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1; minptime=10; stereo=1; sprop-stereo=1
a=rtpmap:110 telephone-event/48000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=fingerprint:sha-256 FD:DF:48:02:C3:F8:D9:57:F8:C8:97:CD:3A:AB:6A:A6:97:4A:BA:A5:48:5A:3A:2D:0F:18:14:EC:D5:A9:21:86
a=setup:active
a=rtcp-mux
a=rtcp:24210 IN IP4 192.168.1.170
a=ice-ufrag:mScCKXydfOjcMntK
a=ice-pwd:Depc4PrcDqdqfgirZlTz8iCe
a=candidate:3496438569 1 udp 659136 192.168.1.170 24210 typ host generation 0
a=end-of-candidates
a=ssrc:3132020181 cname:7pVPJ0AphX8bXYd1
a=ssrc:3132020181 msid:t9mERteVPrwT3aZbexPlp2U6Cspg78dq a0
a=ssrc:3132020181 mslabel:t9mERteVPrwT3aZbexPlp2U6Cspg78dq
a=ssrc:3132020181 label:t9mERteVPrwT3aZbexPlp2U6Cspg78dqa0
m=video 0 UDP/TLS/RTP/SAVPF 19

2020-04-24 14:13:49.395900 [NOTICE] mod_dptools.c:1360 Channel [verto.rtc/5715976] has been answered

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200424/1066e768/attachment-0001.html>

From Jazmin.Marino at on24.com  Fri Apr 24 20:16:09 2020
From: Jazmin.Marino at on24.com (Jazmin Marina Florez Marino)
Date: Fri, 24 Apr 2020 20:16:09 +0000
Subject: [Freeswitch-users] ADD 2 ICE CANDIDATES  mod_verto
Message-ID: <EAA0402198B6144FA20A2B37EB4E1A656E01E3F6@R-ESXMBX01.on24.com>

Hello all,

I am working with mod_verto in a scenario where I need to send public and private IP addresses as ICE candidates.
When I modify the ext-rtp-ip parameter, it only sends one option as typ host ice candidate. However, for my implementation I need to send both.

What I need is the combination of those ice candidates, something like this:
a=candidate:8228103366 1 udp 659136 190.84.118.88 31028 typ host generation 0
a=candidate:3496438569 1 udp 659136 192.168.1.170 24210 typ host generation 0

Is it possible to add both ice candidates in the same SDP from configuration files?

Thanks
Jazmin Florez Marino
+57 304 564 0193
Jazmin.marino at on24.com

SDP with local IP
2020-04-24 14:13:49.395900 [DEBUG] mod_verto.c:2502 Local SDP verto.rtc/5715976:
v=0
o=FreeSWITCH 1587731419 1587731420 IN IP4 192.168.1.170
s=FreeSWITCH
c=IN IP4 192.168.1.170
t=0 0
a=msid-semantic: WMS t9mERteVPrwT3aZbexPlp2U6Cspg78dq
m=audio 24210 UDP/TLS/RTP/SAVPF 111 110
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1; minptime=10; stereo=1; sprop-stereo=1
a=rtpmap:110 telephone-event/48000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=fingerprint:sha-256 FD:DF:48:02:C3:F8:D9:57:F8:C8:97:CD:3A:AB:6A:A6:97:4A:BA:A5:48:5A:3A:2D:0F:18:14:EC:D5:A9:21:86
a=setup:active
a=rtcp-mux
a=rtcp:24210 IN IP4 192.168.1.170
a=ice-ufrag:mScCKXydfOjcMntK
a=ice-pwd:Depc4PrcDqdqfgirZlTz8iCe
a=candidate:3496438569 1 udp 659136 192.168.1.170 24210 typ host generation 0
a=end-of-candidates
a=ssrc:3132020181 cname:7pVPJ0AphX8bXYd1
a=ssrc:3132020181 msid:t9mERteVPrwT3aZbexPlp2U6Cspg78dq a0
a=ssrc:3132020181 mslabel:t9mERteVPrwT3aZbexPlp2U6Cspg78dq
a=ssrc:3132020181 label:t9mERteVPrwT3aZbexPlp2U6Cspg78dqa0
m=video 0 UDP/TLS/RTP/SAVPF 19

2020-04-24 14:13:49.395900 [NOTICE] mod_dptools.c:1360 Channel [verto.rtc/5715976] has been answered


SDP with Public IP
<param name="ext-rtp-ip" value="190.84.118.88"/>
2020-04-24 14:17:23.715620 [DEBUG] mod_verto.c:2502 Local SDP verto.rtc/5715976:
v=0
o=FreeSWITCH 1587724815 1587724816 IN IP4 190.84.118.88
s=FreeSWITCH
c=IN IP4 190.84.118.88
t=0 0
a=msid-semantic: WMS 8gumHFWDC0sD12iPvVhSeEhUfLdXQBVO
m=audio 31028 UDP/TLS/RTP/SAVPF 111 110
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1; minptime=10; stereo=1; sprop-stereo=1
a=rtpmap:110 telephone-event/48000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=fingerprint:sha-256 FD:DF:48:02:C3:F8:D9:57:F8:C8:97:CD:3A:AB:6A:A6:97:4A:BA:A5:48:5A:3A:2D:0F:18:14:EC:D5:A9:21:86
a=setup:active
a=rtcp-mux
a=rtcp:31028 IN IP4 190.84.118.88
a=ice-ufrag:BiYFM6pSItFoUq2t
a=ice-pwd:wTz7x9b00ImLonRPvGl8CBRb
a=candidate:8228103366 1 udp 659136 190.84.118.88 31028 typ host generation 0
a=end-of-candidates
a=ssrc:2930644107 cname:If1gLNJdD1HvUPbg
a=ssrc:2930644107 msid:8gumHFWDC0sD12iPvVhSeEhUfLdXQBVO a0
a=ssrc:2930644107 mslabel:8gumHFWDC0sD12iPvVhSeEhUfLdXQBVO
a=ssrc:2930644107 label:8gumHFWDC0sD12iPvVhSeEhUfLdXQBVOa0
m=video 0 UDP/TLS/RTP/SAVPF 19



-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200424/23dd96e8/attachment-0001.html>

From leo.lai.wolf at gmail.com  Sat Apr 25 05:27:33 2020
From: leo.lai.wolf at gmail.com (Leo Wolf)
Date: Sat, 25 Apr 2020 13:27:33 +0800
Subject: [Freeswitch-users] bridge two outbound calls but got different sdp
Message-ID: <CAPvUGqCFWa3tX6v-A4ZM5Kyg=_j7CGghifVOn3YbXLucTGzBxQ@mail.gmail.com>

Hi,

I usr freeswitch to call out A & B, and want to bridge calls after A & B
responded. After bridging the two call, I found out that the sdp codec
negotiated from A and from B are different, e.g., 102 and 111. This will
cause A & B codec not sync problem. Is there any way to force clients using
specific codec in freeswitch?

I have found out some code in switch_core_media.c:

/* it could be 98 but chrome reserves 98 and 99 for some internal stuff
even though they should not.
   Everyone expects dtmf to be at 101 and Its not worth the trouble so
we'll start at 102 */
smh->payload_space = 102;

Seems the codec send to clients always begin from 102?

Many Thanks,
Leo
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200425/c3d5a1a0/attachment.html>

From petedao at gmail.com  Sat Apr 25 20:41:01 2020
From: petedao at gmail.com (Pete Kay)
Date: Sat, 25 Apr 2020 13:41:01 -0700
Subject: [Freeswitch-users] Need help with mod_v8 install
Message-ID: <CAPxXe8V+RZpHFvLSgfiWoqp1rp+muZPrjzP3wPNT8JYEiYb89g@mail.gmail.com>

Hi


I am getting error when trying to install mod_v8.


Could someone pls help?


# make

make  all-am

make[1]: Entering directory `/usr/src/freeswitch/src/mod/languages/mod_v8'

  CXX      mod_v8_la-mod_v8.lo

In file included from *mod_v8.h:36:0*,

                 from *mod_v8.cpp:68*:

*./include/javascript.hpp:35:16:* *fatal error: *v8.h: No such file or
directory

 #include <v8.h>

*                ^*

compilation terminated.

make[1]: *** [mod_v8_la-mod_v8.lo] Error 1

make[1]: Leaving directory `/usr/src/freeswitch/src/mod/languages/mod_v8'

make: *** [all] Error 2
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200425/fe60d427/attachment.html>

From quang at telepro.me  Fri Apr 24 10:39:22 2020
From: quang at telepro.me (Mai Duy Quang)
Date: Fri, 24 Apr 2020 17:39:22 +0700
Subject: [Freeswitch-users] curl zombie_exec
Message-ID: <FABBEDAF-2213-49C4-89A4-F4E8FC53D447@telepro.me>

Hi,

I call curl in lua script but has this error, been googling all day long but no hope. Any have fix for that?

Channel is hungup and application curl does not have the zombie_exec flag.

my lua code:
				-- to allow curl after the call
				session:execute("set_zombie_exec");

				freeswitch.consoleLog("notice", "[curl][ring groups][missed]  " .. missed_call_data .. "  ");

				session:execute("curl", missed_call_data .. " phone=" .. caller_id_number .. "&target=" .. ring_group_name);
				curl_response_code = session:getVariable("curl_response_code")
				curl_response      = session:getVariable("curl_response_data”)


Many thanks,
Johnathan Mai
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200424/0bd65505/attachment.html>

From rangeare.01 at gmail.com  Mon Apr 27 18:00:00 2020
From: rangeare.01 at gmail.com (Jay Desai)
Date: Mon, 27 Apr 2020 23:30:00 +0530
Subject: [Freeswitch-users] Stream File on both leg simultaneously
Message-ID: <CAGvKJUU2yWYKuoYVHh=TDfACODkerLkgHqi6AUrH7stbQeKdBA@mail.gmail.com>

Hi,

Is there a way to stream file to both legs simultaneously? I'm trying to
play a consent message to both parties before recording a call.

I've tried this
https://lists.freeswitch.org/pipermail/freeswitch-users/2014-February/103093.html
without any luck the mux option seems to make no difference. I'm on
FreeSWITCH version (*1.10.2-release+git~20191231T140119Z~f7bdd3845a~64bit*)
and using XML dial plan to execute commands. It plays file alternatively
while the other leg is muted.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200427/fb9655a7/attachment.html>

From reb.blumstein at gmail.com  Sun Apr 26 12:20:46 2020
From: reb.blumstein at gmail.com (Yisroel Meir Blumstein)
Date: Sun, 26 Apr 2020 15:20:46 +0300
Subject: [Freeswitch-users] mod_conference and Hebrew-language conferencing
 (pro-bono production server for school system)
Message-ID: <CAPh1wwx9uUdU_224DiT78_QMn25eX472Nj7HQVsbVWAw+0JyRQ@mail.gmail.com>

We want to set up a production system handling at least 1000 users per
server, but ideally on the order of 100,000 users on one server. This would
temporarily replace a local, or even regional, school system (pro-bono
work, but they'd pay server costs; this will obviously cost a lot less than
they're going to need to pay the more established company).

Most of these people will need to use a Hebrew conference interface, so we
also need to know if freeswitch/src/mod/say/mod_say_he/mod_say_he.c is
production-ready.

In the worst-case-scenario (which is obviously crazy), where everyone is
talking at once, and these people are in anywhere from 10-30 person
conferences (classes), what are the needed server specs?

In a more realistic scenario, where a maximum of 2 people are talking at
once per conference:

   - 20,000 users speaking, 2 per each of the 10,000 conferences
   - ~6700 users speaking, 2 per each of the ~3300 conferences

Or, 3 people at once:

   - 30,000 users speaking, 3 per each of the 10,000 conferences
   - ~10,000 users speaking, 3 per each of the ~33,333 conferences

What are the variables here? My friend runs Bestfone
<https://github.com/signalwire/freeswitch/issues/www.bestfone.com>, and
—while he handles tens of thousands of calls a month— he doesn't handle
that many simultaneous calls, so we're trying to figure out exactly what
level of power is needed here.

Please help! We're trying to literally help thousands of school kids (over
124,527, as of last official numbers) if that motivates you.


Thanks!

<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
Virus-free.
www.avast.com
<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
<#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200426/025e7a76/attachment-0001.html>

From seanmhansen7 at gmail.com  Sat Apr 25 18:30:52 2020
From: seanmhansen7 at gmail.com (Sean Hansen)
Date: Sat, 25 Apr 2020 11:30:52 -0700
Subject: [Freeswitch-users] sofia reject MESSAGE method
In-Reply-To: <CADMnEC-B64+xPjK9hrmoTcPP_2ZTb-N-49zMoqfUMg4GTUWs9A@mail.gmail.com>
References: <CAF=KPsNJDr0CjhVpRs0qKZnsigGKoTX95Dw5=n_tc3wy2gFvZg@mail.gmail.com>
 <CADMnEC-B64+xPjK9hrmoTcPP_2ZTb-N-49zMoqfUMg4GTUWs9A@mail.gmail.com>
Message-ID: <CAF=KPsPDey1xiCj397G_vm1s+WStjTeZ6vVXV8S8NKVyM6n54g@mail.gmail.com>

Thanks Loi,

This was the solution.

On Tue, Apr 21, 2020 at 9:29 PM Loi Dang <loidang at hoiio.com> wrote:

> Hi, try setting this var `enable-chat` to false in your sofia sip profile.
>
> rgds,
> Loi Dang
>
> On Wed, Apr 22, 2020 at 10:32 AM Sean Hansen <seanmhansen7 at gmail.com>
> wrote:
>
>> Hello,
>>
>> Does anyone know of a way to reject chat/messaging for sofia?
>> Specifically the MESSAGE method?
>>
>> We use FreeSWITCH purely as an SBC/media proxy for voice only, so
>> therefore have no need for directory/chatplan/etc. We have noticed with our
>> current configuration, when our PBX passes a MESSAGE (usually generated on
>> accident...) to freeswitch it sends the MESSAGE right back to the PBX -
>> which creates a loop and jams up the proxies.
>>
>> The log is complaining as such:
>> freeswitch.log.2020-04-21-11-57-04.1:2020-04-21 10:50:43.195480 [DEBUG]
>> sofia_presence.c:225 Can't find registered user XXXXXXXXXXX at domain.com
>> freeswitch.log.2020-04-21-11-57-04.1:2020-04-21 10:50:43.195480 [DEBUG]
>> sofia_presence.c:225 Can't find registered user XXXXXXXXXXX at domain.com
>> freeswitch.log.2020-04-21-11-57-04.1:2020-04-21 10:50:43.215485 [DEBUG]
>> sofia_presence.c:225 Can't find registered user XXXXXXXXXXX at domain.com
>> freeswitch.log.2020-04-21-11-57-04.1:2020-04-21 10:50:43.215485 [DEBUG]
>> sofia_presence.c:225 Can't find registered user XXXXXXXXXXX at domain.com
>>
>> where XXXXXXXXXXX is the destination number for the message.
>>
>> We do not load mod_sms, and all of the chatplans are disabled.
>>
>> Thanks.
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200425/3cc48429/attachment.html>

From youssef.elouam at outlook.com  Mon Apr 27 08:42:41 2020
From: youssef.elouam at outlook.com (Youssef ELOUAM)
Date: Mon, 27 Apr 2020 08:42:41 +0000
Subject: [Freeswitch-users] send message via python script using ESL library
Message-ID: <VI1PR0101MB2286B20661EE16E611459E978DAF0@VI1PR0101MB2286.eurprd01.prod.exchangelabs.com>

Hi everyone,

I am trying to send SMS via python script but not working, I followed the script in confluence

from ESL import *

con = ESLconnection("127.0.0.1", "8021", "ClueCon")

if con.connected():
    event = ESLevent("CUSTOM", "SMS::SEND_MESSAGE")

    event.addHeader("to", "1000 at 192.168.1.6")
    event.addHeader("from", "5000 at 192.168.1.6")
    event.addHeader("dest_proto", "sip")

    print(event.serialize("plain"))
    event.addBody("message contents")
    con.sendEvent(event)


as result of print command I get :


Event-Name: CUSTOM

Event-Subclass: SMS%3A%3ASEND_MESSAGE

to: 1000%40192.168.1.6

from: 5000%40192.168.1.6

dest_proto: sip

but no message received in 1000 extension.


Youssef ELOUAM
https://www.linkedin.com/in/elouam/

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200427/63b803cb/attachment-0001.html>

From kaduww at gmail.com  Mon Apr 27 19:56:43 2020
From: kaduww at gmail.com (Carlos Eduardo)
Date: Mon, 27 Apr 2020 16:56:43 -0300
Subject: [Freeswitch-users] Stream File on both leg simultaneously
In-Reply-To: <CAGvKJUU2yWYKuoYVHh=TDfACODkerLkgHqi6AUrH7stbQeKdBA@mail.gmail.com>
References: <CAGvKJUU2yWYKuoYVHh=TDfACODkerLkgHqi6AUrH7stbQeKdBA@mail.gmail.com>
Message-ID: <CAB=EceZw4XayK5_gUefyp=NVVoOZBtRuccf-=sNp46_KnBzp8A@mail.gmail.com>

Hello Jay,

Meybe the API command uuid_broadcast wuld help

"Execute an arbitrary dialplan application, typically playing a media file,
on a specific uuid. If a filename is specified then it is played into the
channel(s). To execute an application use "app::args" syntax."
uuid_broadcast <uuid> <path> [aleg|bleg|both]

You can call this API after the call gets answered, using the
execute_on_answer variable


Em seg., 27 de abr. de 2020 às 16:42, Jay Desai <rangeare.01 at gmail.com>
escreveu:

> Hi,
>
> Is there a way to stream file to both legs simultaneously? I'm trying to
> play a consent message to both parties before recording a call.
>
> I've tried this
> https://lists.freeswitch.org/pipermail/freeswitch-users/2014-February/103093.html
> without any luck the mux option seems to make no difference. I'm on
> FreeSWITCH version (*1.10.2-release+git~20191231T140119Z~f7bdd3845a~64bit*)
> and using XML dial plan to execute commands. It plays file alternatively
> while the other leg is muted.
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
*Carlos E. Wagner*
*Tecnólogo em Telecomunicações, Opensips Certified Professional*

*Fone: +55 48 99981-0894*
*E-mail:* kaduww at gmail.com
*LinkedIn:* https://www.linkedin.com/in/carlos-eduardo-wagner-96bbb433/
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200427/bdb042cd/attachment.html>

From max at nysolutions.com  Mon Apr 27 21:24:57 2020
From: max at nysolutions.com (Moishe Grunstein)
Date: Mon, 27 Apr 2020 21:24:57 +0000
Subject: [Freeswitch-users] mod_conference and Hebrew-language
 conferencing (pro-bono production server for school system)
In-Reply-To: <CAPh1wwx9uUdU_224DiT78_QMn25eX472Nj7HQVsbVWAw+0JyRQ@mail.gmail.com>
References: <CAPh1wwx9uUdU_224DiT78_QMn25eX472Nj7HQVsbVWAw+0JyRQ@mail.gmail.com>
Message-ID: <46043da93dfa41cd931b0d2c23592861@nysolutions.com>

Will depend much more on your hardware then anything else, why don’t you split into multiple systems.

Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US
Service Request Email: support at nysolutions.com<mailto:support at nysolutions.com>
[cid:image001.jpg at 01C72F94.9EE45D60]<http://www.nysolutions.com/>
Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Yisroel Meir Blumstein
Sent: Sunday, April 26, 2020 8:21 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] mod_conference and Hebrew-language conferencing (pro-bono production server for school system)


We want to set up a production system handling at least 1000 users per server, but ideally on the order of 100,000 users on one server. This would temporarily replace a local, or even regional, school system (pro-bono work, but they'd pay server costs; this will obviously cost a lot less than they're going to need to pay the more established company).

Most of these people will need to use a Hebrew conference interface, so we also need to know if freeswitch/src/mod/say/mod_say_he/mod_say_he.c is production-ready.

In the worst-case-scenario (which is obviously crazy), where everyone is talking at once, and these people are in anywhere from 10-30 person conferences (classes), what are the needed server specs?

In a more realistic scenario, where a maximum of 2 people are talking at once per conference:

  *   20,000 users speaking, 2 per each of the 10,000 conferences
  *   ~6700 users speaking, 2 per each of the ~3300 conferences

Or, 3 people at once:

  *   30,000 users speaking, 3 per each of the 10,000 conferences
  *   ~10,000 users speaking, 3 per each of the ~33,333 conferences

What are the variables here? My friend runs Bestfone<https://github.com/signalwire/freeswitch/issues/www.bestfone.com>, and —while he handles tens of thousands of calls a month— he doesn't handle that many simultaneous calls, so we're trying to figure out exactly what level of power is needed here.

Please help! We're trying to literally help thousands of school kids (over 124,527, as of last official numbers) if that motivates you.



Thanks!

[https://ipmcdn.avast.com/images/icons/icon-envelope-tick-round-orange-animated-no-repeat-v1.gif]<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>

Virus-free. www.avast.com<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>


-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200427/882bcdd4/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 2424 bytes
Desc: image001.jpg
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200427/882bcdd4/attachment-0001.jpg>

From brian at freeswitch.com  Tue Apr 28 00:06:36 2020
From: brian at freeswitch.com (Brian West)
Date: Mon, 27 Apr 2020 19:06:36 -0500
Subject: [Freeswitch-users] sofia reject MESSAGE method
In-Reply-To: <CAF=KPsPDey1xiCj397G_vm1s+WStjTeZ6vVXV8S8NKVyM6n54g@mail.gmail.com>
References: <CAF=KPsNJDr0CjhVpRs0qKZnsigGKoTX95Dw5=n_tc3wy2gFvZg@mail.gmail.com>
 <CADMnEC-B64+xPjK9hrmoTcPP_2ZTb-N-49zMoqfUMg4GTUWs9A@mail.gmail.com>
 <CAF=KPsPDey1xiCj397G_vm1s+WStjTeZ6vVXV8S8NKVyM6n54g@mail.gmail.com>
Message-ID: <CAEJMVkBzs0x4yhZVED198AKFfLz0aUzEFrQj6Jb+c2n37Ls1gQ@mail.gmail.com>

Thats a param not a variable, variables have underscores and params have
dashes, just to clarify.

On Mon, Apr 27, 2020 at 4:16 PM Sean Hansen <seanmhansen7 at gmail.com> wrote:

> Thanks Loi,
>
> This was the solution.
>
> On Tue, Apr 21, 2020 at 9:29 PM Loi Dang <loidang at hoiio.com> wrote:
>
>> Hi, try setting this var `enable-chat` to false in your sofia sip profile.
>>
>> rgds,
>> Loi Dang
>>
>> On Wed, Apr 22, 2020 at 10:32 AM Sean Hansen <seanmhansen7 at gmail.com>
>> wrote:
>>
>>> Hello,
>>>
>>> Does anyone know of a way to reject chat/messaging for sofia?
>>> Specifically the MESSAGE method?
>>>
>>> We use FreeSWITCH purely as an SBC/media proxy for voice only, so
>>> therefore have no need for directory/chatplan/etc. We have noticed with our
>>> current configuration, when our PBX passes a MESSAGE (usually generated on
>>> accident...) to freeswitch it sends the MESSAGE right back to the PBX -
>>> which creates a loop and jams up the proxies.
>>>
>>> The log is complaining as such:
>>> freeswitch.log.2020-04-21-11-57-04.1:2020-04-21 10:50:43.195480 [DEBUG]
>>> sofia_presence.c:225 Can't find registered user XXXXXXXXXXX at domain.com
>>> freeswitch.log.2020-04-21-11-57-04.1:2020-04-21 10:50:43.195480 [DEBUG]
>>> sofia_presence.c:225 Can't find registered user XXXXXXXXXXX at domain.com
>>> freeswitch.log.2020-04-21-11-57-04.1:2020-04-21 10:50:43.215485 [DEBUG]
>>> sofia_presence.c:225 Can't find registered user XXXXXXXXXXX at domain.com
>>> freeswitch.log.2020-04-21-11-57-04.1:2020-04-21 10:50:43.215485 [DEBUG]
>>> sofia_presence.c:225 Can't find registered user XXXXXXXXXXX at domain.com
>>>
>>> where XXXXXXXXXXX is the destination number for the message.
>>>
>>> We do not load mod_sms, and all of the chatplans are disabled.
>>>
>>> Thanks.
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 

Brian West | Co-founder and Developer

Need Commercial support? email sales at freeswitch.com

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

Email: brian at freeswitch.com

Mobile: 918-424-9378

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

[image: https://www.facebook.com/signalwireinc?src=email]
<https://www.facebook.com/freeswitch> [image:
https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200427/a20386fa/attachment.html>

From loidang at hoiio.com  Tue Apr 28 02:50:42 2020
From: loidang at hoiio.com (Loi Dang)
Date: Tue, 28 Apr 2020 09:50:42 +0700
Subject: [Freeswitch-users] sofia reject MESSAGE method
In-Reply-To: <CAEJMVkBzs0x4yhZVED198AKFfLz0aUzEFrQj6Jb+c2n37Ls1gQ@mail.gmail.com>
References: <CAF=KPsNJDr0CjhVpRs0qKZnsigGKoTX95Dw5=n_tc3wy2gFvZg@mail.gmail.com>
 <CADMnEC-B64+xPjK9hrmoTcPP_2ZTb-N-49zMoqfUMg4GTUWs9A@mail.gmail.com>
 <CAF=KPsPDey1xiCj397G_vm1s+WStjTeZ6vVXV8S8NKVyM6n54g@mail.gmail.com>
 <CAEJMVkBzs0x4yhZVED198AKFfLz0aUzEFrQj6Jb+c2n37Ls1gQ@mail.gmail.com>
Message-ID: <CADMnEC9rXmKxZdZaVWLv_N9yprikcZ6e0EVibgRx=O=QMgTVfw@mail.gmail.com>

`variables have underscores and params have dashes` good point to notice,
thanks Brian.

rgds,
Loi Dang

On Tue, Apr 28, 2020 at 7:31 AM Brian West <brian at freeswitch.com> wrote:

> Thats a param not a variable, variables have underscores and params have
> dashes, just to clarify.
>
> On Mon, Apr 27, 2020 at 4:16 PM Sean Hansen <seanmhansen7 at gmail.com>
> wrote:
>
>> Thanks Loi,
>>
>> This was the solution.
>>
>> On Tue, Apr 21, 2020 at 9:29 PM Loi Dang <loidang at hoiio.com> wrote:
>>
>>> Hi, try setting this var `enable-chat` to false in your sofia sip
>>> profile.
>>>
>>> rgds,
>>> Loi Dang
>>>
>>> On Wed, Apr 22, 2020 at 10:32 AM Sean Hansen <seanmhansen7 at gmail.com>
>>> wrote:
>>>
>>>> Hello,
>>>>
>>>> Does anyone know of a way to reject chat/messaging for sofia?
>>>> Specifically the MESSAGE method?
>>>>
>>>> We use FreeSWITCH purely as an SBC/media proxy for voice only, so
>>>> therefore have no need for directory/chatplan/etc. We have noticed with our
>>>> current configuration, when our PBX passes a MESSAGE (usually generated on
>>>> accident...) to freeswitch it sends the MESSAGE right back to the PBX -
>>>> which creates a loop and jams up the proxies.
>>>>
>>>> The log is complaining as such:
>>>> freeswitch.log.2020-04-21-11-57-04.1:2020-04-21 10:50:43.195480 [DEBUG]
>>>> sofia_presence.c:225 Can't find registered user XXXXXXXXXXX at domain.com
>>>> freeswitch.log.2020-04-21-11-57-04.1:2020-04-21 10:50:43.195480 [DEBUG]
>>>> sofia_presence.c:225 Can't find registered user XXXXXXXXXXX at domain.com
>>>> freeswitch.log.2020-04-21-11-57-04.1:2020-04-21 10:50:43.215485 [DEBUG]
>>>> sofia_presence.c:225 Can't find registered user XXXXXXXXXXX at domain.com
>>>> freeswitch.log.2020-04-21-11-57-04.1:2020-04-21 10:50:43.215485 [DEBUG]
>>>> sofia_presence.c:225 Can't find registered user XXXXXXXXXXX at domain.com
>>>>
>>>> where XXXXXXXXXXX is the destination number for the message.
>>>>
>>>> We do not load mod_sms, and all of the chatplans are disabled.
>>>>
>>>> Thanks.
>>>>
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
>
> Brian West | Co-founder and Developer
>
> Need Commercial support? email sales at freeswitch.com
>
> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>
> Email: brian at freeswitch.com
>
> Mobile: 918-424-9378
>
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>
> [image: https://www.facebook.com/signalwireinc?src=email]
> <https://www.facebook.com/freeswitch> [image:
> https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200428/2ac66f31/attachment-0001.html>

From s.safarov at gmail.com  Tue Apr 28 06:12:59 2020
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 28 Apr 2020 09:12:59 +0300
Subject: [Freeswitch-users] mod_conference and Hebrew-language
 conferencing (pro-bono production server for school system)
In-Reply-To: <46043da93dfa41cd931b0d2c23592861@nysolutions.com>
References: <CAPh1wwx9uUdU_224DiT78_QMn25eX472Nj7HQVsbVWAw+0JyRQ@mail.gmail.com>
 <46043da93dfa41cd931b0d2c23592861@nysolutions.com>
Message-ID: <CAHtxdDd=T535pfeGhXMT0MEUEV+_0CoOQ1bGz4VXQcjUZHeCrw@mail.gmail.com>

Also you make FreeSwitch mod_conference load tests on Amazon server.
When you get production ready results then you can answer your question.

On Tue, Apr 28, 2020 at 12:38 AM Moishe Grunstein <max at nysolutions.com>
wrote:

> Will depend much more on your hardware then anything else, why don’t you
> split into multiple systems.
>
>
>
> Thanks,
>
>
>
> Moishe Grunstein
>
> Tornado Computer Systems, Inc.
>
> 212.400.7650 888.IPPBX.US
> *Service Request Email: support at nysolutions.com <support at nysolutions.com> *
>
> [image: cid:image001.jpg at 01C72F94.9EE45D60] <http://www.nysolutions.com/>
>
> Computer Networking * Managed Services * IP Video Surveillance * Network
> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network
> Security * Site Surveys * CMS
>
>
>
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *On
> Behalf Of *Yisroel Meir Blumstein
> *Sent:* Sunday, April 26, 2020 8:21 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] mod_conference and Hebrew-language
> conferencing (pro-bono production server for school system)
>
>
>
> We want to set up a production system handling at least 1000 users per
> server, but ideally on the order of 100,000 users on one server. This would
> temporarily replace a local, or even regional, school system (pro-bono
> work, but they'd pay server costs; this will obviously cost a lot less than
> they're going to need to pay the more established company).
>
> Most of these people will need to use a Hebrew conference interface, so we
> also need to know if freeswitch/src/mod/say/mod_say_he/mod_say_he.c is
> production-ready.
>
> In the worst-case-scenario (which is obviously crazy), where everyone is
> talking at once, and these people are in anywhere from 10-30 person
> conferences (classes), what are the needed server specs?
>
> In a more realistic scenario, where a maximum of 2 people are talking at
> once per conference:
>
>    - 20,000 users speaking, 2 per each of the 10,000 conferences
>    - ~6700 users speaking, 2 per each of the ~3300 conferences
>
> Or, 3 people at once:
>
>    - 30,000 users speaking, 3 per each of the 10,000 conferences
>    - ~10,000 users speaking, 3 per each of the ~33,333 conferences
>
> What are the variables here? My friend runs Bestfone
> <https://github.com/signalwire/freeswitch/issues/www.bestfone.com>, and
> —while he handles tens of thousands of calls a month— he doesn't handle
> that many simultaneous calls, so we're trying to figure out exactly what
> level of power is needed here.
>
> Please help! We're trying to literally help thousands of school kids (over
> 124,527, as of last official numbers) if that motivates you.
>
>
>
> Thanks!
>
>
>
>
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
>
> Virus-free. www.avast.com
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200428/6bf2c4a8/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 2424 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200428/6bf2c4a8/attachment.jpg>

From dujinfang at gmail.com  Tue Apr 28 08:22:18 2020
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 28 Apr 2020 16:22:18 +0800
Subject: [Freeswitch-users] curl zombie_exec
In-Reply-To: <FABBEDAF-2213-49C4-89A4-F4E8FC53D447@telepro.me>
References: <FABBEDAF-2213-49C4-89A4-F4E8FC53D447@telepro.me>
Message-ID: <CAJs+-5rHr=O79AJUMVTEHGoxBUF12jPtDffKQ1Kks8=bUTgQ=Q@mail.gmail.com>

you have to patch mod_curl do to that like below. please make a pr if it
works for you.

SWITCH_ADD_APP(app_interface, "curl", "Perform a http request", "Perform a
http request",
curl_app_function, SYNTAX, SAF_SUPPORT_NOMEDIA | SAF_ROUTING_EXEC |
SAF_ZOMBIE_EXEC);


On Tue, Apr 28, 2020 at 4:07 AM Mai Duy Quang <quang at telepro.me> wrote:

> Hi,
>
> I call curl in lua script but has this error, been googling all day long
> but no hope. Any have fix for that?
>
> Channel is hungup and application curl does not have the zombie_exec flag.
>
> my lua code:
> -- to allow curl after the call
> session:execute("set_zombie_exec");
>
> freeswitch.consoleLog("notice", "[curl][ring groups][missed]  " ..
> missed_call_data .. "  ");
>
> session:execute("curl", missed_call_data .. " phone=" .. caller_id_number
> .. "&target=" .. ring_group_name);
> curl_response_code = session:getVariable("curl_response_code")
> curl_response      = session:getVariable("curl_response_data”)
>
>
> Many thanks,
> Johnathan Mai
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200428/5b48c11d/attachment-0001.html>

From dujinfang at gmail.com  Tue Apr 28 08:27:03 2020
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 28 Apr 2020 16:27:03 +0800
Subject: [Freeswitch-users] Announcing SaraPhone,
 SIP WebRTC Open Source business phone
In-Reply-To: <CAAoGpGSzph5NMAmDNzLgvQrG2_w7zLVi8UG7UXYnf6M2O_sPqQ@mail.gmail.com>
References: <CALXCt0qaDgo0gSmkKSy0uzK+5v3tS6KRJgXN78u2qFWL6JH4MA@mail.gmail.com>
 <CAAoGpGSzph5NMAmDNzLgvQrG2_w7zLVi8UG7UXYnf6M2O_sPqQ@mail.gmail.com>
Message-ID: <CAJs+-5ovMbpnLz7g4Erivh_QdrFOqzZEZNy8euCAha14sTZcKQ@mail.gmail.com>

Nice ;)

On Fri, Apr 17, 2020 at 2:31 AM Mitul Limbani <mitul at enterux.com> wrote:

> Great Giovanni, shall definitely try this in our environment :-)
>
> Mitul Limbani
>
> On Thu, 16 Apr, 2020, 21:05 Giovanni Maruzzelli, <gmaruzz at gmail.com>
> wrote:
>
>> My fellow VoIPers,
>>
>> I am pleased to announce the early availability of:
>>
>> SaraPhone
>> ------------------
>>
>> SaraPhone is a bare bone SIP WebRTC voice phone, complete with most
>> features real companies want to use in real world: HotDesking, Redial,
>> BLFs, MWI, DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer,
>> Notifications, running on all Browsers both on Desktop and SmartPhone.
>>
>> SaraPhone is fully integrated with FusionPBX, the full-featured domain
>> based multi-tenant PBX and voice switch for FreeSwitch.
>>
>> Based on SIP.js, SaraPhone works with all WebRTC compliant SIP proxies,
>> gateways, and servers (Asterisk, OpenSIPS, Kamailio, etc).
>>
>> Initial author is Giovanni Maruzzelli, and SaraPhone gets its name from
>> Giovanni's wife, Sara Hosseini.
>>
>> In addition to providing all of the usual DeskPhone functionality,
>> SaraPhone got:
>>
>>    - Desktop Notification for Incoming Calls
>>    - Live MWI update
>>    - Real Time BLFs status update
>>    - BLF click to call
>>    - Caller Name and Number Display
>>    - Call Error Cause Display
>>    - AutoAnswer
>>    - Network Disconnect Reload
>>    - Show and Set Caller-ID (incoming-outbound)
>>
>>
>> You an find it in GitHub ( https://github.com/gmaruzz/saraphone ).
>>
>> Anyone interested can play with it :).
>>
>> Have fun,
>> giovanni
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> OpenTelecom.IT
>> cell: +39 347 266 56 18
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200428/8b907855/attachment.html>

From dujinfang at gmail.com  Tue Apr 28 08:33:15 2020
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 28 Apr 2020 16:33:15 +0800
Subject: [Freeswitch-users] freeswitch video conference recording black
	screen
In-Reply-To: <CAPXdxKh3DuZq47-suVZYCe-QTooAEsE2iz-6V9iy_HLZ2oKtLA@mail.gmail.com>
References: <CAPXdxKh3DuZq47-suVZYCe-QTooAEsE2iz-6V9iy_HLZ2oKtLA@mail.gmail.com>
Message-ID: <CAJs+-5rroJkHTUQnhvVh7qH8zNeD5sXR+9EM4=_zNGa=yTsXZA@mail.gmail.com>

Did the first caller into the conference have video?

On Wed, Apr 22, 2020 at 11:42 AM Aidar Kamalov <aidar.kamalov at gmail.com>
wrote:

> Hello, I am trying to record video conference:
> <param name="auto-record"
> value="$${recordings_dir}/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.mp4"/>
>
> But I am getting black screen with audio only in recording file. Although
> I can record video on every single channels:
> uuid_record UUID start /var/lib/freeswitch/recordings/UUID.mp4
>
> --
> Aydar A. Kamalov
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200428/9ea83012/attachment.html>

From gmaruzz at gmail.com  Tue Apr 28 09:23:38 2020
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 28 Apr 2020 11:23:38 +0200
Subject: [Freeswitch-users] Announcing SaraPhone,
 SIP WebRTC Open Source business phone
In-Reply-To: <CAJs+-5ovMbpnLz7g4Erivh_QdrFOqzZEZNy8euCAha14sTZcKQ@mail.gmail.com>
References: <CALXCt0qaDgo0gSmkKSy0uzK+5v3tS6KRJgXN78u2qFWL6JH4MA@mail.gmail.com>
 <CAAoGpGSzph5NMAmDNzLgvQrG2_w7zLVi8UG7UXYnf6M2O_sPqQ@mail.gmail.com>
 <CAJs+-5ovMbpnLz7g4Erivh_QdrFOqzZEZNy8euCAha14sTZcKQ@mail.gmail.com>
Message-ID: <CALXCt0o5f758qjP24A9y1z_KAKOGcUz9kaTsyA8fo4k+2tt8nw@mail.gmail.com>

Seven <3



On Tue, Apr 28, 2020 at 11:17 AM Seven Du <dujinfang at gmail.com> wrote:

> Nice ;)
>
> On Fri, Apr 17, 2020 at 2:31 AM Mitul Limbani <mitul at enterux.com> wrote:
>
>> Great Giovanni, shall definitely try this in our environment :-)
>>
>> Mitul Limbani
>>
>> On Thu, 16 Apr, 2020, 21:05 Giovanni Maruzzelli, <gmaruzz at gmail.com>
>> wrote:
>>
>>> My fellow VoIPers,
>>>
>>> I am pleased to announce the early availability of:
>>>
>>> SaraPhone
>>> ------------------
>>>
>>> SaraPhone is a bare bone SIP WebRTC voice phone, complete with most
>>> features real companies want to use in real world: HotDesking, Redial,
>>> BLFs, MWI, DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer,
>>> Notifications, running on all Browsers both on Desktop and SmartPhone.
>>>
>>> SaraPhone is fully integrated with FusionPBX, the full-featured domain
>>> based multi-tenant PBX and voice switch for FreeSwitch.
>>>
>>> Based on SIP.js, SaraPhone works with all WebRTC compliant SIP proxies,
>>> gateways, and servers (Asterisk, OpenSIPS, Kamailio, etc).
>>>
>>> Initial author is Giovanni Maruzzelli, and SaraPhone gets its name from
>>> Giovanni's wife, Sara Hosseini.
>>>
>>> In addition to providing all of the usual DeskPhone functionality,
>>> SaraPhone got:
>>>
>>>    - Desktop Notification for Incoming Calls
>>>    - Live MWI update
>>>    - Real Time BLFs status update
>>>    - BLF click to call
>>>    - Caller Name and Number Display
>>>    - Call Error Cause Display
>>>    - AutoAnswer
>>>    - Network Disconnect Reload
>>>    - Show and Set Caller-ID (incoming-outbound)
>>>
>>>
>>> You an find it in GitHub ( https://github.com/gmaruzz/saraphone ).
>>>
>>> Anyone interested can play with it :).
>>>
>>> Have fun,
>>> giovanni
>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> OpenTelecom.IT
>>> cell: +39 347 266 56 18
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200428/92e14ea7/attachment-0001.html>

From babak.freeswitch at gmail.com  Wed Apr 29 07:09:31 2020
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Wed, 29 Apr 2020 11:39:31 +0430
Subject: [Freeswitch-users] Acl to ignore mdns ice candidates generated by
	browsers
Message-ID: <CAEVfcaagbW+Osv9VmYLb75BskiDtaL5sCZTLrzkc8V-kOsebMQ@mail.gmail.com>

Hi
I'm trying to use sipjs with freeswitch but the problem is freeswitch is
accepting candidates like this:
a=candidate:2131708102 1 udp 2113937151
ad7f0d0b-7245-4c33-84af-fda3f2921d18.local 54385 typ host generation 0
network-cost 999
these are generated by mdns from browser and cause errors like
 AUDIO RTP REPORTS ERROR: [Remote Address Error!]
and then
INCOMPATIBLE_DESTINATION

I tried wan_v4.auto as "apply-candidate-acl" but it is accepting those
candidate too
is there any builtin acl that I can use to make fs accept only public ipv4
addresses?
thanks
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200429/b4b36677/attachment.html>

From covici at ccs.covici.com  Wed Apr 29 08:14:03 2020
From: covici at ccs.covici.com (John Covici)
Date: Wed, 29 Apr 2020 04:14:03 -0400
Subject: [Freeswitch-users] Announcing SaraPhone,
	SIP WebRTC Open Source business phone
In-Reply-To: <CALXCt0o5f758qjP24A9y1z_KAKOGcUz9kaTsyA8fo4k+2tt8nw@mail.gmail.com>
References: <CALXCt0qaDgo0gSmkKSy0uzK+5v3tS6KRJgXN78u2qFWL6JH4MA@mail.gmail.com>
 <CAAoGpGSzph5NMAmDNzLgvQrG2_w7zLVi8UG7UXYnf6M2O_sPqQ@mail.gmail.com>
 <CAJs+-5ovMbpnLz7g4Erivh_QdrFOqzZEZNy8euCAha14sTZcKQ@mail.gmail.com>
 <CALXCt0o5f758qjP24A9y1z_KAKOGcUz9kaTsyA8fo4k+2tt8nw@mail.gmail.com>
Message-ID: <m3v9li4sl0.wl-covici@ccs.covici.com>

If I were to install this, any way to protect so not anyone could use
it?  Also, any plans to get video so I could use it with video
conference?

On Tue, 28 Apr 2020 05:23:38 -0400,
Giovanni Maruzzelli wrote:
> 
> [1  <multipart/alternative (7bit)>]
> [1.1  <text/plain; UTF-8 (7bit)>]
> Seven <3
> 
> 
> 
> On Tue, Apr 28, 2020 at 11:17 AM Seven Du <dujinfang at gmail.com> wrote:
> 
> > Nice ;)
> >
> > On Fri, Apr 17, 2020 at 2:31 AM Mitul Limbani <mitul at enterux.com> wrote:
> >
> >> Great Giovanni, shall definitely try this in our environment :-)
> >>
> >> Mitul Limbani
> >>
> >> On Thu, 16 Apr, 2020, 21:05 Giovanni Maruzzelli, <gmaruzz at gmail.com>
> >> wrote:
> >>
> >>> My fellow VoIPers,
> >>>
> >>> I am pleased to announce the early availability of:
> >>>
> >>> SaraPhone
> >>> ------------------
> >>>
> >>> SaraPhone is a bare bone SIP WebRTC voice phone, complete with most
> >>> features real companies want to use in real world: HotDesking, Redial,
> >>> BLFs, MWI, DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer,
> >>> Notifications, running on all Browsers both on Desktop and SmartPhone.
> >>>
> >>> SaraPhone is fully integrated with FusionPBX, the full-featured domain
> >>> based multi-tenant PBX and voice switch for FreeSwitch.
> >>>
> >>> Based on SIP.js, SaraPhone works with all WebRTC compliant SIP proxies,
> >>> gateways, and servers (Asterisk, OpenSIPS, Kamailio, etc).
> >>>
> >>> Initial author is Giovanni Maruzzelli, and SaraPhone gets its name from
> >>> Giovanni's wife, Sara Hosseini.
> >>>
> >>> In addition to providing all of the usual DeskPhone functionality,
> >>> SaraPhone got:
> >>>
> >>>    - Desktop Notification for Incoming Calls
> >>>    - Live MWI update
> >>>    - Real Time BLFs status update
> >>>    - BLF click to call
> >>>    - Caller Name and Number Display
> >>>    - Call Error Cause Display
> >>>    - AutoAnswer
> >>>    - Network Disconnect Reload
> >>>    - Show and Set Caller-ID (incoming-outbound)
> >>>
> >>>
> >>> You an find it in GitHub ( https://github.com/gmaruzz/saraphone ).
> >>>
> >>> Anyone interested can play with it :).
> >>>
> >>> Have fun,
> >>> giovanni
> >>>
> >>>
> >>> --
> >>> Sincerely,
> >>>
> >>> Giovanni Maruzzelli
> >>> OpenTelecom.IT
> >>> cell: +39 347 266 56 18
> >>>
> >>> _________________________________________________________________________
> >>>
> >>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> >>> services.
> >>> Build your next product on our scalable cloud platform.
> >>>
> >>> Join our online community to chat in real time
> >>> https://signalwire.community
> >>>
> >>> Professional FreeSWITCH Services
> >>> sales at freeswitch.com
> >>> https://freeswitch.com
> >>>
> >>> Official FreeSWITCH Sites
> >>> https://freeswitch.com/oss
> >>> https://freeswitch.org/confluence
> >>> https://cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> https://freeswitch.com
> >>
> >> _________________________________________________________________________
> >>
> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> >> services.
> >> Build your next product on our scalable cloud platform.
> >>
> >> Join our online community to chat in real time
> >> https://signalwire.community
> >>
> >> Professional FreeSWITCH Services
> >> sales at freeswitch.com
> >> https://freeswitch.com
> >>
> >> Official FreeSWITCH Sites
> >> https://freeswitch.com/oss
> >> https://freeswitch.org/confluence
> >> https://cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> https://freeswitch.com
> >
> >
> >
> > --
> > About: http://about.me/dujinfang
> > Blog: http://www.dujinfang.com
> > Proj:  http://www.freeswitch.org.cn
> > _________________________________________________________________________
> >
> > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> > services.
> > Build your next product on our scalable cloud platform.
> >
> > Join our online community to chat in real time
> > https://signalwire.community
> >
> > Professional FreeSWITCH Services
> > sales at freeswitch.com
> > https://freeswitch.com
> >
> > Official FreeSWITCH Sites
> > https://freeswitch.com/oss
> > https://freeswitch.org/confluence
> > https://cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > https://freeswitch.com
> 
> 
> 
> -- 
> Sincerely,
> 
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
> [1.2  <text/html; UTF-8 (quoted-printable)>]
> [2  <text/plain; utf-8 (base64)>]
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici wb2una
         covici at ccs.covici.com


From gmaruzz at gmail.com  Wed Apr 29 08:49:31 2020
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 29 Apr 2020 10:49:31 +0200
Subject: [Freeswitch-users] Announcing SaraPhone,
 SIP WebRTC Open Source business phone
In-Reply-To: <m3v9li4sl0.wl-covici@ccs.covici.com>
References: <CALXCt0qaDgo0gSmkKSy0uzK+5v3tS6KRJgXN78u2qFWL6JH4MA@mail.gmail.com>
 <CAAoGpGSzph5NMAmDNzLgvQrG2_w7zLVi8UG7UXYnf6M2O_sPqQ@mail.gmail.com>
 <CAJs+-5ovMbpnLz7g4Erivh_QdrFOqzZEZNy8euCAha14sTZcKQ@mail.gmail.com>
 <CALXCt0o5f758qjP24A9y1z_KAKOGcUz9kaTsyA8fo4k+2tt8nw@mail.gmail.com>
 <m3v9li4sl0.wl-covici@ccs.covici.com>
Message-ID: <CALXCt0qAXmJ2H3=1tPON5F16_L=SQnzNMV4ZJ5L+egtq+-amwQ@mail.gmail.com>

On Wed, Apr 29, 2020 at 10:35 AM John Covici <covici at ccs.covici.com> wrote:

> If I were to install this, any way to protect so not anyone could use
>

Hello John,
Maybe I have not understood the question, but SaraPhone live in the
browser, and connect via SSL to HTTPS server, and via WSS to SIP server.

So, standard techniques apply.

SIP account is protected by usual login/password, that travel encrypted WSS
Webpage interaction is encrypted SSL

You can further protect it all by ACL in FreeSWITCH and webserver, by using
client side SSL certificates, etc... Also, you can restrict IP port access
via firewall to known sources...



> it?  Also, any plans to get video so I could use it with video
> conference?
>
>
No, SaraPhone is a business phone, intended as a desktop phone
replacement/impersonator/hotdesk/roamingdesk

In future I will release a video client with all bells and whistles for
collaboration and videoconferencing.

The use cases, requirements and feature sets are very different, one only
client will be bad serving both cases (in my opinion and experience).

Have a nice continuation,

-giovanni








> On Tue, 28 Apr 2020 05:23:38 -0400,
> Giovanni Maruzzelli wrote:
> >
> > [1  <multipart/alternative (7bit)>]
> > [1.1  <text/plain; UTF-8 (7bit)>]
> > Seven <3
> >
> >
> >
> > On Tue, Apr 28, 2020 at 11:17 AM Seven Du <dujinfang at gmail.com> wrote:
> >
> > > Nice ;)
> > >
> > > On Fri, Apr 17, 2020 at 2:31 AM Mitul Limbani <mitul at enterux.com>
> wrote:
> > >
> > >> Great Giovanni, shall definitely try this in our environment :-)
> > >>
> > >> Mitul Limbani
> > >>
> > >> On Thu, 16 Apr, 2020, 21:05 Giovanni Maruzzelli, <gmaruzz at gmail.com>
> > >> wrote:
> > >>
> > >>> My fellow VoIPers,
> > >>>
> > >>> I am pleased to announce the early availability of:
> > >>>
> > >>> SaraPhone
> > >>> ------------------
> > >>>
> > >>> SaraPhone is a bare bone SIP WebRTC voice phone, complete with most
> > >>> features real companies want to use in real world: HotDesking,
> Redial,
> > >>> BLFs, MWI, DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer,
> > >>> Notifications, running on all Browsers both on Desktop and
> SmartPhone.
> > >>>
> > >>> SaraPhone is fully integrated with FusionPBX, the full-featured
> domain
> > >>> based multi-tenant PBX and voice switch for FreeSwitch.
> > >>>
> > >>> Based on SIP.js, SaraPhone works with all WebRTC compliant SIP
> proxies,
> > >>> gateways, and servers (Asterisk, OpenSIPS, Kamailio, etc).
> > >>>
> > >>> Initial author is Giovanni Maruzzelli, and SaraPhone gets its name
> from
> > >>> Giovanni's wife, Sara Hosseini.
> > >>>
> > >>> In addition to providing all of the usual DeskPhone functionality,
> > >>> SaraPhone got:
> > >>>
> > >>>    - Desktop Notification for Incoming Calls
> > >>>    - Live MWI update
> > >>>    - Real Time BLFs status update
> > >>>    - BLF click to call
> > >>>    - Caller Name and Number Display
> > >>>    - Call Error Cause Display
> > >>>    - AutoAnswer
> > >>>    - Network Disconnect Reload
> > >>>    - Show and Set Caller-ID (incoming-outbound)
> > >>>
> > >>>
> > >>> You an find it in GitHub ( https://github.com/gmaruzz/saraphone ).
> > >>>
> > >>> Anyone interested can play with it :).
> > >>>
> > >>> Have fun,
> > >>> giovanni
> > >>>
> > >>>
> > >>> --
> > >>> Sincerely,
> > >>>
> > >>> Giovanni Maruzzelli
> > >>> OpenTelecom.IT
> > >>> cell: +39 347 266 56 18
> > >>>
> > >>>
> _________________________________________________________________________
> > >>>
> > >>> The FreeSWITCH project is sponsored by SignalWire
> https://signalwire.com
> > >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> > >>> services.
> > >>> Build your next product on our scalable cloud platform.
> > >>>
> > >>> Join our online community to chat in real time
> > >>> https://signalwire.community
> > >>>
> > >>> Professional FreeSWITCH Services
> > >>> sales at freeswitch.com
> > >>> https://freeswitch.com
> > >>>
> > >>> Official FreeSWITCH Sites
> > >>> https://freeswitch.com/oss
> > >>> https://freeswitch.org/confluence
> > >>> https://cluecon.com
> > >>>
> > >>> FreeSWITCH-users mailing list
> > >>> FreeSWITCH-users at lists.freeswitch.org
> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >>> https://freeswitch.com
> > >>
> > >>
> _________________________________________________________________________
> > >>
> > >> The FreeSWITCH project is sponsored by SignalWire
> https://signalwire.com
> > >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> > >> services.
> > >> Build your next product on our scalable cloud platform.
> > >>
> > >> Join our online community to chat in real time
> > >> https://signalwire.community
> > >>
> > >> Professional FreeSWITCH Services
> > >> sales at freeswitch.com
> > >> https://freeswitch.com
> > >>
> > >> Official FreeSWITCH Sites
> > >> https://freeswitch.com/oss
> > >> https://freeswitch.org/confluence
> > >> https://cluecon.com
> > >>
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> https://freeswitch.com
> > >
> > >
> > >
> > > --
> > > About: http://about.me/dujinfang
> > > Blog: http://www.dujinfang.com
> > > Proj:  http://www.freeswitch.org.cn
> > >
> _________________________________________________________________________
> > >
> > > The FreeSWITCH project is sponsored by SignalWire
> https://signalwire.com
> > > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> > > services.
> > > Build your next product on our scalable cloud platform.
> > >
> > > Join our online community to chat in real time
> > > https://signalwire.community
> > >
> > > Professional FreeSWITCH Services
> > > sales at freeswitch.com
> > > https://freeswitch.com
> > >
> > > Official FreeSWITCH Sites
> > > https://freeswitch.com/oss
> > > https://freeswitch.org/confluence
> > > https://cluecon.com
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > https://freeswitch.com
> >
> >
> >
> > --
> > Sincerely,
> >
> > Giovanni Maruzzelli
> > OpenTelecom.IT
> > cell: +39 347 266 56 18
> > [1.2  <text/html; UTF-8 (quoted-printable)>]
> > [2  <text/plain; utf-8 (base64)>]
> > _________________________________________________________________________
> >
> > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> > Build your next product on our scalable cloud platform.
> >
> > Join our online community to chat in real time
> https://signalwire.community
> >
> > Professional FreeSWITCH Services
> > sales at freeswitch.com
> > https://freeswitch.com
> >
> > Official FreeSWITCH Sites
> > https://freeswitch.com/oss
> > https://freeswitch.org/confluence
> > https://cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > https://freeswitch.com
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>          John Covici wb2una
>          covici at ccs.covici.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200429/92dd1045/attachment-0001.html>

From asilva at wirelessmundi.com  Wed Apr 29 10:07:29 2020
From: asilva at wirelessmundi.com (=?utf-8?Q?Ant=C3=B3nio_Silva?=)
Date: Wed, 29 Apr 2020 12:07:29 +0200
Subject: mod_verto: subscribe events for specific accounts
In-Reply-To: <4DC90D2D-6FF9-4985-99F0-48A0565BFEDD@wirelessmundi.com>
References: <4DC90D2D-6FF9-4985-99F0-48A0565BFEDD@wirelessmundi.com>
Message-ID: <FF06CB65-8A7E-4BDA-8807-87806ABC31AC@wirelessmundi.com>

Reppling to myself, just found the right syntax,  it is posible from verto.js:

To get all events:
vertoHandle.subscribe(“presene”, {
			handler: function(v, e) {
				log.debug("***presence event e:", e);
			}
		});

But to get specific events per account:
vertoHandle.subscribe([“presene.ACCOUNT1”, “presene.ACCOUNT2”], {
			handler: function(v, e) {
				log.debug("***presence event e:", e);
			}
		});

Hope it help others :)


> On 17 Mar 2020, at 13:53, António Silva <asilva at wirelessmundi.com> wrote:
> 
> Hi,
> 
> Is it possible to subscribe using mod_verto?
> 
> Right now I’ve enable in mod_verto the presence events but I get all the events for all the accounts, and is fine but i was wondering if it’s is possible to just do a single subscribe like in sip.
> 
> Thanks for the help.
> 
> 
> Regards,
> António



From gregor at infomedia.si  Wed Apr 29 10:23:32 2020
From: gregor at infomedia.si (Gregor Nanger)
Date: Wed, 29 Apr 2020 12:23:32 +0200
Subject: [Freeswitch-users] mod_verto: subscribe events for specific
	accounts
In-Reply-To: <mailman.2724.1588154883.1226.freeswitch-users@lists.freeswitch.org>
References: <4DC90D2D-6FF9-4985-99F0-48A0565BFEDD@wirelessmundi.com>
 <mailman.2724.1588154883.1226.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAC7ttFF1z5DJcpUoTGTowZ4cmvxkW1PQw+Tbht3uVk2CaXae2g@mail.gmail.com>

Thanx for sharing. Could be usefull.

One question. Is this right syntax:
vertoHandle.subscribe(“presene”, {

Is it presene or presence?

Best regards, Gregor

On Wed, 29 Apr 2020 at 12:08, António Silva via FreeSWITCH-users <
freeswitch-users at lists.freeswitch.org> wrote:

>
>
>
> ---------- Forwarded message ----------
> From: "António Silva" <asilva at wirelessmundi.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Wed, 29 Apr 2020 12:07:29 +0200
> Subject: Re: mod_verto: subscribe events for specific accounts
> Reppling to myself, just found the right syntax,  it is posible from
> verto.js:
>
> To get all events:
> vertoHandle.subscribe(“presene”, {
>                         handler: function(v, e) {
>                                 log.debug("***presence event e:", e);
>                         }
>                 });
>
> But to get specific events per account:
> vertoHandle.subscribe([“presene.ACCOUNT1”, “presene.ACCOUNT2”], {
>                         handler: function(v, e) {
>                                 log.debug("***presence event e:", e);
>                         }
>                 });
>
> Hope it help others :)
>
>
> > On 17 Mar 2020, at 13:53, António Silva <asilva at wirelessmundi.com>
> wrote:
> >
> > Hi,
> >
> > Is it possible to subscribe using mod_verto?
> >
> > Right now I’ve enable in mod_verto the presence events but I get all the
> events for all the accounts, and is fine but i was wondering if it’s is
> possible to just do a single subscribe like in sip.
> >
> > Thanks for the help.
> >
> >
> > Regards,
> > António
>
>
>
>
>
> ---------- Forwarded message ----------
> From: "António Silva via FreeSWITCH-users" <
> freeswitch-users at lists.freeswitch.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Wed, 29 Apr 2020 03:08:12 -0700 (PDT)
> Subject: Re: [Freeswitch-users] mod_verto: subscribe events for specific
> accounts
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
• Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
• www.infomedia.si
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200429/a1706bee/attachment.html>

From covici at ccs.covici.com  Wed Apr 29 14:19:23 2020
From: covici at ccs.covici.com (John Covici)
Date: Wed, 29 Apr 2020 10:19:23 -0400
Subject: [Freeswitch-users] Announcing SaraPhone,
	SIP WebRTC Open Source business phone
In-Reply-To: <CALXCt0qAXmJ2H3=1tPON5F16_L=SQnzNMV4ZJ5L+egtq+-amwQ@mail.gmail.com>
References: <CALXCt0qaDgo0gSmkKSy0uzK+5v3tS6KRJgXN78u2qFWL6JH4MA@mail.gmail.com>
 <CAAoGpGSzph5NMAmDNzLgvQrG2_w7zLVi8UG7UXYnf6M2O_sPqQ@mail.gmail.com>
 <CAJs+-5ovMbpnLz7g4Erivh_QdrFOqzZEZNy8euCAha14sTZcKQ@mail.gmail.com>
 <CALXCt0o5f758qjP24A9y1z_KAKOGcUz9kaTsyA8fo4k+2tt8nw@mail.gmail.com>
 <m3v9li4sl0.wl-covici@ccs.covici.com>
 <CALXCt0qAXmJ2H3=1tPON5F16_L=SQnzNMV4ZJ5L+egtq+-amwQ@mail.gmail.com>
Message-ID: <m3tv124bo4.wl-covici@ccs.covici.com>

OK, thanks.

On Wed, 29 Apr 2020 04:49:31 -0400,
Giovanni Maruzzelli wrote:
> 
> [1  <multipart/alternative (7bit)>]
> [1.1  <text/plain; UTF-8 (7bit)>]
> On Wed, Apr 29, 2020 at 10:35 AM John Covici <covici at ccs.covici.com> wrote:
> 
> > If I were to install this, any way to protect so not anyone could use
> >
> 
> Hello John,
> Maybe I have not understood the question, but SaraPhone live in the
> browser, and connect via SSL to HTTPS server, and via WSS to SIP server.
> 
> So, standard techniques apply.
> 
> SIP account is protected by usual login/password, that travel encrypted WSS
> Webpage interaction is encrypted SSL
> 
> You can further protect it all by ACL in FreeSWITCH and webserver, by using
> client side SSL certificates, etc... Also, you can restrict IP port access
> via firewall to known sources...
> 
> 
> 
> > it?  Also, any plans to get video so I could use it with video
> > conference?
> >
> >
> No, SaraPhone is a business phone, intended as a desktop phone
> replacement/impersonator/hotdesk/roamingdesk
> 
> In future I will release a video client with all bells and whistles for
> collaboration and videoconferencing.
> 
> The use cases, requirements and feature sets are very different, one only
> client will be bad serving both cases (in my opinion and experience).
> 
> Have a nice continuation,
> 
> -giovanni
> 
> 
> 
> 
> 
> 
> 
> 
> > On Tue, 28 Apr 2020 05:23:38 -0400,
> > Giovanni Maruzzelli wrote:
> > >
> > > [1  <multipart/alternative (7bit)>]
> > > [1.1  <text/plain; UTF-8 (7bit)>]
> > > Seven <3
> > >
> > >
> > >
> > > On Tue, Apr 28, 2020 at 11:17 AM Seven Du <dujinfang at gmail.com> wrote:
> > >
> > > > Nice ;)
> > > >
> > > > On Fri, Apr 17, 2020 at 2:31 AM Mitul Limbani <mitul at enterux.com>
> > wrote:
> > > >
> > > >> Great Giovanni, shall definitely try this in our environment :-)
> > > >>
> > > >> Mitul Limbani
> > > >>
> > > >> On Thu, 16 Apr, 2020, 21:05 Giovanni Maruzzelli, <gmaruzz at gmail.com>
> > > >> wrote:
> > > >>
> > > >>> My fellow VoIPers,
> > > >>>
> > > >>> I am pleased to announce the early availability of:
> > > >>>
> > > >>> SaraPhone
> > > >>> ------------------
> > > >>>
> > > >>> SaraPhone is a bare bone SIP WebRTC voice phone, complete with most
> > > >>> features real companies want to use in real world: HotDesking,
> > Redial,
> > > >>> BLFs, MWI, DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer,
> > > >>> Notifications, running on all Browsers both on Desktop and
> > SmartPhone.
> > > >>>
> > > >>> SaraPhone is fully integrated with FusionPBX, the full-featured
> > domain
> > > >>> based multi-tenant PBX and voice switch for FreeSwitch.
> > > >>>
> > > >>> Based on SIP.js, SaraPhone works with all WebRTC compliant SIP
> > proxies,
> > > >>> gateways, and servers (Asterisk, OpenSIPS, Kamailio, etc).
> > > >>>
> > > >>> Initial author is Giovanni Maruzzelli, and SaraPhone gets its name
> > from
> > > >>> Giovanni's wife, Sara Hosseini.
> > > >>>
> > > >>> In addition to providing all of the usual DeskPhone functionality,
> > > >>> SaraPhone got:
> > > >>>
> > > >>>    - Desktop Notification for Incoming Calls
> > > >>>    - Live MWI update
> > > >>>    - Real Time BLFs status update
> > > >>>    - BLF click to call
> > > >>>    - Caller Name and Number Display
> > > >>>    - Call Error Cause Display
> > > >>>    - AutoAnswer
> > > >>>    - Network Disconnect Reload
> > > >>>    - Show and Set Caller-ID (incoming-outbound)
> > > >>>
> > > >>>
> > > >>> You an find it in GitHub ( https://github.com/gmaruzz/saraphone ).
> > > >>>
> > > >>> Anyone interested can play with it :).
> > > >>>
> > > >>> Have fun,
> > > >>> giovanni
> > > >>>
> > > >>>
> > > >>> --
> > > >>> Sincerely,
> > > >>>
> > > >>> Giovanni Maruzzelli
> > > >>> OpenTelecom.IT
> > > >>> cell: +39 347 266 56 18
> > > >>>
> > > >>>
> > _________________________________________________________________________
> > > >>>
> > > >>> The FreeSWITCH project is sponsored by SignalWire
> > https://signalwire.com
> > > >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> > > >>> services.
> > > >>> Build your next product on our scalable cloud platform.
> > > >>>
> > > >>> Join our online community to chat in real time
> > > >>> https://signalwire.community
> > > >>>
> > > >>> Professional FreeSWITCH Services
> > > >>> sales at freeswitch.com
> > > >>> https://freeswitch.com
> > > >>>
> > > >>> Official FreeSWITCH Sites
> > > >>> https://freeswitch.com/oss
> > > >>> https://freeswitch.org/confluence
> > > >>> https://cluecon.com
> > > >>>
> > > >>> FreeSWITCH-users mailing list
> > > >>> FreeSWITCH-users at lists.freeswitch.org
> > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >>> UNSUBSCRIBE:
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > >>> https://freeswitch.com
> > > >>
> > > >>
> > _________________________________________________________________________
> > > >>
> > > >> The FreeSWITCH project is sponsored by SignalWire
> > https://signalwire.com
> > > >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> > > >> services.
> > > >> Build your next product on our scalable cloud platform.
> > > >>
> > > >> Join our online community to chat in real time
> > > >> https://signalwire.community
> > > >>
> > > >> Professional FreeSWITCH Services
> > > >> sales at freeswitch.com
> > > >> https://freeswitch.com
> > > >>
> > > >> Official FreeSWITCH Sites
> > > >> https://freeswitch.com/oss
> > > >> https://freeswitch.org/confluence
> > > >> https://cluecon.com
> > > >>
> > > >> FreeSWITCH-users mailing list
> > > >> FreeSWITCH-users at lists.freeswitch.org
> > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >> UNSUBSCRIBE:
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > >> https://freeswitch.com
> > > >
> > > >
> > > >
> > > > --
> > > > About: http://about.me/dujinfang
> > > > Blog: http://www.dujinfang.com
> > > > Proj:  http://www.freeswitch.org.cn
> > > >
> > _________________________________________________________________________
> > > >
> > > > The FreeSWITCH project is sponsored by SignalWire
> > https://signalwire.com
> > > > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> > > > services.
> > > > Build your next product on our scalable cloud platform.
> > > >
> > > > Join our online community to chat in real time
> > > > https://signalwire.community
> > > >
> > > > Professional FreeSWITCH Services
> > > > sales at freeswitch.com
> > > > https://freeswitch.com
> > > >
> > > > Official FreeSWITCH Sites
> > > > https://freeswitch.com/oss
> > > > https://freeswitch.org/confluence
> > > > https://cluecon.com
> > > >
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > https://freeswitch.com
> > >
> > >
> > >
> > > --
> > > Sincerely,
> > >
> > > Giovanni Maruzzelli
> > > OpenTelecom.IT
> > > cell: +39 347 266 56 18
> > > [1.2  <text/html; UTF-8 (quoted-printable)>]
> > > [2  <text/plain; utf-8 (base64)>]
> > > _________________________________________________________________________
> > >
> > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> > > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> > services.
> > > Build your next product on our scalable cloud platform.
> > >
> > > Join our online community to chat in real time
> > https://signalwire.community
> > >
> > > Professional FreeSWITCH Services
> > > sales at freeswitch.com
> > > https://freeswitch.com
> > >
> > > Official FreeSWITCH Sites
> > > https://freeswitch.com/oss
> > > https://freeswitch.org/confluence
> > > https://cluecon.com
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > https://freeswitch.com
> >
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> >
> >          John Covici wb2una
> >          covici at ccs.covici.com
> >
> > _________________________________________________________________________
> >
> > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> > services.
> > Build your next product on our scalable cloud platform.
> >
> > Join our online community to chat in real time
> > https://signalwire.community
> >
> > Professional FreeSWITCH Services
> > sales at freeswitch.com
> > https://freeswitch.com
> >
> > Official FreeSWITCH Sites
> > https://freeswitch.com/oss
> > https://freeswitch.org/confluence
> > https://cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > https://freeswitch.com
> 
> 
> 
> -- 
> Sincerely,
> 
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
> [1.2  <text/html; UTF-8 (quoted-printable)>]
> [2  <text/plain; utf-8 (base64)>]
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici wb2una
         covici at ccs.covici.com


From brian at freeswitch.com  Wed Apr 29 20:17:13 2020
From: brian at freeswitch.com (Brian West)
Date: Wed, 29 Apr 2020 15:17:13 -0500
Subject: [Freeswitch-users] ADD 2 ICE CANDIDATES mod_verto
In-Reply-To: <EAA0402198B6144FA20A2B37EB4E1A656E01E3E2@R-ESXMBX01.on24.com>
References: <EAA0402198B6144FA20A2B37EB4E1A656E01E3E2@R-ESXMBX01.on24.com>
Message-ID: <CAEJMVkAvu1zP_Tpjm1d9uqfrcyKzp=QJuF203XhtE1XxThzF7g@mail.gmail.com>

set the variable include_external_ip=true

/b

On Mon, Apr 27, 2020 at 3:02 PM Jazmin Marina Florez Marino <
Jazmin.Marino at on24.com> wrote:

> Hello all,
>
>
>
> I am working with mod_verto in a scenario where I need to send public and
> private IP addresses as ICE candidates.
>
> When I modify the ext-rtp-ip parameter, it only sends one option as typ
> host ice candidate. However, for my implementation I need to send both.
>
>
>
> What I need is the combination of those ice candidates, something like
> this:
>
> a=candidate:8228103366 1 udp 659136 190.84.118.88 31028 typ host
> generation 0
>
> a=candidate:3496438569 1 udp 659136 192.168.1.170 24210 typ host
> generation 0
>
>
>
> Is it possible to add both ice candidates in the same SDP from
> configuration files?
>
>
>
>
>
> *SDP with local IP*
>
> 2020-04-24 14:17:23.715620 [DEBUG] mod_verto.c:2502 Local SDP
> verto.rtc/5715976:
>
> v=0
>
> o=FreeSWITCH 1587724815 1587724816 IN IP4 190.84.118.88
>
> s=FreeSWITCH
>
> c=IN IP4 190.84.118.88
>
> t=0 0
>
> a=msid-semantic: WMS 8gumHFWDC0sD12iPvVhSeEhUfLdXQBVO
>
> m=audio 31028 UDP/TLS/RTP/SAVPF 111 110
>
> a=rtpmap:111 opus/48000/2
>
> a=fmtp:111 useinbandfec=1; minptime=10; stereo=1; sprop-stereo=1
>
> a=rtpmap:110 telephone-event/48000
>
> a=silenceSupp:off - - - -
>
> a=ptime:20
>
> a=sendrecv
>
> a=fingerprint:sha-256
> FD:DF:48:02:C3:F8:D9:57:F8:C8:97:CD:3A:AB:6A:A6:97:4A:BA:A5:48:5A:3A:2D:0F:18:14:EC:D5:A9:21:86
>
> a=setup:active
>
> a=rtcp-mux
>
> a=rtcp:31028 IN IP4 190.84.118.88
>
> a=ice-ufrag:BiYFM6pSItFoUq2t
>
> a=ice-pwd:wTz7x9b00ImLonRPvGl8CBRb
>
> *a=candidate:8228103366 1 udp 659136 190.84.118.88 31028 typ host
> generation 0*
>
> a=end-of-candidates
>
> a=ssrc:2930644107 cname:If1gLNJdD1HvUPbg
>
> a=ssrc:2930644107 msid:8gumHFWDC0sD12iPvVhSeEhUfLdXQBVO a0
>
> a=ssrc:2930644107 mslabel:8gumHFWDC0sD12iPvVhSeEhUfLdXQBVO
>
> a=ssrc:2930644107 label:8gumHFWDC0sD12iPvVhSeEhUfLdXQBVOa0
>
> m=video 0 UDP/TLS/RTP/SAVPF 19
>
>
>
> *SDP with Public IP*
>
> *<param name="ext-rtp-ip" value="190.84.118.88"/>*
>
> 2020-04-24 14:13:49.395900 [DEBUG] mod_verto.c:2502 Local SDP
> verto.rtc/5715976:
>
> v=0
>
> o=FreeSWITCH 1587731419 1587731420 IN IP4 192.168.1.170
>
> s=FreeSWITCH
>
> c=IN IP4 192.168.1.170
>
> t=0 0
>
> a=msid-semantic: WMS t9mERteVPrwT3aZbexPlp2U6Cspg78dq
>
> m=audio 24210 UDP/TLS/RTP/SAVPF 111 110
>
> a=rtpmap:111 opus/48000/2
>
> a=fmtp:111 useinbandfec=1; minptime=10; stereo=1; sprop-stereo=1
>
> a=rtpmap:110 telephone-event/48000
>
> a=silenceSupp:off - - - -
>
> a=ptime:20
>
> a=sendrecv
>
> a=fingerprint:sha-256
> FD:DF:48:02:C3:F8:D9:57:F8:C8:97:CD:3A:AB:6A:A6:97:4A:BA:A5:48:5A:3A:2D:0F:18:14:EC:D5:A9:21:86
>
> a=setup:active
>
> a=rtcp-mux
>
> a=rtcp:24210 IN IP4 192.168.1.170
>
> a=ice-ufrag:mScCKXydfOjcMntK
>
> a=ice-pwd:Depc4PrcDqdqfgirZlTz8iCe
>
> *a=candidate:3496438569 1 udp 659136 192.168.1.170 24210 typ host
> generation 0*
>
> a=end-of-candidates
>
> a=ssrc:3132020181 cname:7pVPJ0AphX8bXYd1
>
> a=ssrc:3132020181 msid:t9mERteVPrwT3aZbexPlp2U6Cspg78dq a0
>
> a=ssrc:3132020181 mslabel:t9mERteVPrwT3aZbexPlp2U6Cspg78dq
>
> a=ssrc:3132020181 label:t9mERteVPrwT3aZbexPlp2U6Cspg78dqa0
>
> m=video 0 UDP/TLS/RTP/SAVPF 19
>
>
>
> 2020-04-24 14:13:49.395900 [NOTICE] mod_dptools.c:1360 Channel
> [verto.rtc/5715976] has been answered
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 

Brian West | Co-founder and Developer

Need Commercial support? email sales at freeswitch.com

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

Email: brian at freeswitch.com

Mobile: 918-424-9378

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

[image: https://www.facebook.com/signalwireinc?src=email]
<https://www.facebook.com/freeswitch> [image:
https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200429/8af56461/attachment-0001.html>

From davidswalkabout at gmail.com  Wed Apr 29 22:07:06 2020
From: davidswalkabout at gmail.com (David P)
Date: Wed, 29 Apr 2020 15:07:06 -0700
Subject: [Freeswitch-users] TURN TCP to reduce TURN cost
In-Reply-To: <mailman.2733.1588191476.1226.freeswitch-users@lists.freeswitch.org>
References: <mailman.2733.1588191476.1226.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAE23truEnYcwi4k=mdbvMROmGPt9Oprb+9FMR06fVR4dObA6mg@mail.gmail.com>

Brian West <brian at freeswitch.com> wrote:

> Subject: Re: [Freeswitch-users] ADD 2 ICE CANDIDATES mod_verto
> set the variable include_external_ip=true
>

Hey Brian,

Your tip about include_external_ip seemed relevant to my recent question
whether FS supports TURN TCP; that is, can FS be configured to provide a
tcp candidate in its answer sdp, which would save the cost of sending calls
through TURN for users whose firewalls block needed udp ports.

I searched for your variable in the source and found
https://github.com/signalwire/freeswitch/blob/53cd06900c5ce827b2c0ac74485607d9a07d4a7d/src/switch_core_media.c#L9957
which shows that the transport is not frozen as "udp" but is the value of
ice_out->cands[0][0].transport

Is it possible for this transport member to be "tcp"? How could one
configure FS to provide both udp and tcp candidates?

Cheers,
David
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200429/076471a5/attachment.html>

From josen.figueroa at unixmexico.org  Wed Apr 29 23:09:22 2020
From: josen.figueroa at unixmexico.org (Jose Figueroa)
Date: Wed, 29 Apr 2020 18:09:22 -0500
Subject: [Freeswitch-users] Error on loading ESL.so on debian 10
Message-ID: <CAB_jsRjYFdV0k+z4stOy1G1yt7KVtVr8pq+YKVgkmHKDs2gW1Q@mail.gmail.com>

Hello team,

Using:
PHP 7.3.14-1~deb10u1 (cli) (built: Feb 16 2020 15:07:23) ( NTS ) on debian
10
Operating System: Debian GNU/Linux 10 (buster)
Kernel: Linux 4.19.0-6-amd64

I already built de freeswitch v1.10 from your github repository and I want
to use PHP ESL, but I can't make it working, once the ESL.so is built and I
added on php.ini the enable_dl = On I do the php -f ESL.php and it does not
give any output.

But when I do php -f test.php it gives me the following error:

$ php -f test.php
php: symbol lookup error: /usr/lib/php/20180731/ESL.so: undefined symbol:
_ZN13ESLconnectionC1EPKcS1_S1_

What am I doing wrong?

Thanks in advance
JF
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200429/b33dc2bc/attachment.html>

From brian at freeswitch.com  Thu Apr 30 01:59:25 2020
From: brian at freeswitch.com (Brian West)
Date: Wed, 29 Apr 2020 20:59:25 -0500
Subject: [Freeswitch-users] TURN TCP to reduce TURN cost
In-Reply-To: <CAE23truEnYcwi4k=mdbvMROmGPt9Oprb+9FMR06fVR4dObA6mg@mail.gmail.com>
References: <mailman.2733.1588191476.1226.freeswitch-users@lists.freeswitch.org>
 <CAE23truEnYcwi4k=mdbvMROmGPt9Oprb+9FMR06fVR4dObA6mg@mail.gmail.com>
Message-ID: <CAEJMVkBrAjFu4DE3=6a7sHrmK5gW2vW16mfWPYaqYVpTB5gwmQ@mail.gmail.com>

You use relay, you speak TCP to the relay and it turns it into UDP,
FreeSWITCH is oblivious that its even taking place.

On Wed, Apr 29, 2020 at 5:34 PM David P <davidswalkabout at gmail.com> wrote:

> Brian West <brian at freeswitch.com> wrote:
>
>> Subject: Re: [Freeswitch-users] ADD 2 ICE CANDIDATES mod_verto
>> set the variable include_external_ip=true
>>
>
> Hey Brian,
>
> Your tip about include_external_ip seemed relevant to my recent question
> whether FS supports TURN TCP; that is, can FS be configured to provide a
> tcp candidate in its answer sdp, which would save the cost of sending calls
> through TURN for users whose firewalls block needed udp ports.
>
> I searched for your variable in the source and found
>
> https://github.com/signalwire/freeswitch/blob/53cd06900c5ce827b2c0ac74485607d9a07d4a7d/src/switch_core_media.c#L9957
> which shows that the transport is not frozen as "udp" but is the value of
> ice_out->cands[0][0].transport
>
> Is it possible for this transport member to be "tcp"? How could one
> configure FS to provide both udp and tcp candidates?
>
> Cheers,
> David
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 

Brian West | Co-founder and Developer

Need Commercial support? email sales at freeswitch.com

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

Email: brian at freeswitch.com

Mobile: 918-424-9378

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

[image: https://www.facebook.com/signalwireinc?src=email]
<https://www.facebook.com/freeswitch> [image:
https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200429/96891361/attachment.html>

From avi at avimarcus.net  Thu Apr 30 09:12:08 2020
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 30 Apr 2020 09:12:08 +0000
Subject: [Freeswitch-users] FYI - new Amazon Web Services TTS option
In-Reply-To: <9a4e58be-dfd7-781c-0f60-12fedd498167@mst.edu>
References: <10bc193c-6018-5d0e-7628-72d329f4e1b8@mst.edu>
 <CANsY4D+FoS-jM3jUDYZUaS+3S52z9EV5NUuMZTtf6ThbbGL6=A@mail.gmail.com>
 <9a4e58be-dfd7-781c-0f60-12fedd498167@mst.edu>
Message-ID: <01000171ca5b8672-e6a37648-0dec-4364-b40a-553f36c7d285-000000@email.amazonses.com>

I started building a proof of concept for Polly sounds for English. I
thought that Matthew neural version sounded pretty good, and we don't have
a Male voice (just 3x female voices).

https://github.com/avimar/freeswitch-sounds-polly

I don't have it automated to generate a whole set, but there's a bunch of
samples.
Also, can someone please chime in on downsampling... I assume we want
direct PCM but that's only available up to 16000. (And you need to add a
header to make it a normal readable wav)
Mp3 an ogg are available at 24000. Not sure what the best way to get our
samples are, I used sox and got a bunch of versions -- 8000 pcm direct from
amazon, 16000, 16000 downsampled from the ogg....

I also haven't tested mod_say for numbers and currencies to see how it
sounds... probably need to do something to make it flow better.

-Avi Marcus
BestFone

On Thu, Dec 1, 2016 at 4:30 PM Nathan Neulinger <nneul at mst.edu> wrote:

> I'm going to try and put together a sounds set from the freeswitch phrase
> file and will submit back to project.
>
> In my case, even if the word-to-word transitions are not as uniform as
> callie, it will allow my custom prompts to have
> the same voice as the system prompts.
>
> -- Nathan
>
> On 12/01/2016 04:58 AM, Vladyslav Zakhozhai wrote:
> > Hi Nathan,
> >
> > thank you for sharing this aws option. Looks very interesting.
> >
> > 2016-12-01 4:27 GMT+02:00 Nathan Neulinger <nneul at mst.edu <mailto:
> nneul at mst.edu>>:
> >
> >
> >
> https://aws.amazon.com/blogs/aws/polly-text-to-speech-in-47-voices-and-24-languages/
> >     <
> https://aws.amazon.com/blogs/aws/polly-text-to-speech-in-47-voices-and-24-languages/
> >
> >
> >     The quality blows away everything I've tried out previously...
> Highly relevant option for voice prompts.
> >
> >     -- Nathan
> >
> >     ------------------------------------------------------------
> >     Nathan Neulinger nneul at mst.edu <mailto:nneul at mst.edu>
> >     Missouri S&T Information Technology    (573) 612-1412
> >     System Administrator - Architect
> >
> >
>  _________________________________________________________________________
> >     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
> >
> >
> >
> >
> > --
> > С уважением,
> > Владислав Захожай
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> --
> ------------------------------------------------------------
> Nathan Neulinger                       nneul 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200430/67b52313/attachment-0001.html>

From avi at avimarcus.net  Thu Apr 30 09:49:33 2020
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 30 Apr 2020 09:49:33 +0000
Subject: [Freeswitch-users] How to play a file to conference moderator?
Message-ID: <01000171ca7dc8de-e4646598-37a5-46f7-9adc-0df16b2ff7d4-000000@email.amazonses.com>

I'm setting up a conference system for a school, and a feature is instead
of simply unmuting, the student raises their hand.

I hacked together something with execute_application javascript that saves
to coredb sqlite, so we can get a timestamp. (modifying to add a flag
wouldn't be good enough, we want to save the order).

Then I want to play a notice only to the moderator that someone has a
question.
I set up something super-hacky, and was wondering if there's a more robust
way. Or a simple code change to add this functionality (I'm not a C coder,
but it wasn't obvious to add "saymember moderator X" without a bunch more
code)

Since I need the moderator's member id to the conference, which isn't set
until after he joins, I did this:

<action application="set" data="api_result=${jsrun
conference/setModerator.js myConference ${uuid}}" />
<action application="conference" data=" myConference at school
++flags{endconf|moderator}"/>

Then setModerator.js does a uuid get var to get his id:

var memberID = apiExecute("uuid_getvar", argv[1] + " conference_member_id");
console_log("DEBUG","Got ID @ "+i+": " + memberID);
memberID = parseInt(memberID,10);
if(memberID && memberID>0){ //NaN is falsy, we're good.
var saveString="insert/conference_"+argv[0]+"/moderator/"+memberID;
console_log("DEBUG","Done, saving: " + memberID + ", string:"+saveString);
apiExecute("hash",saveString);

I put in a wait and repeat, but I guess starting up v8 takes long enough
that the variable is set by then.

This seems very hacky, is there a better way to do this?

Thanks,
-Avi Marcus
1-718-989-9485 (USA)
02-372-1570 (Israel)
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200430/3e2d1b12/attachment.html>

From francesco at delagarda.com  Thu Apr 30 10:21:46 2020
From: francesco at delagarda.com (Francesco Facco de Lagarda)
Date: Thu, 30 Apr 2020 12:21:46 +0200
Subject: [Freeswitch-users] Calls per gateway
Message-ID: <002d01d61ed9$286f2f00$794d8d00$@delagarda.com>

Hello everyone .. I’ve been dredging the ‘net for a week trying to find a solution but can’t find one.

 

I have a complicated system with 3 2n gateways with sim cards.

Two have been dedicated to national calls and the last to international calls (the SIMS have been purchased with special deals according to the destination)

I just need a simple method to find out how many current calls are on each gateway so I can route my calls accordingly. I use ESL to route my calls.

I’ve tried all the variations of sofia status, show calls, show channels, but I cant find any that just tell me how many calls each gateway is handling at present (in and out)

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200430/e7ea2b01/attachment.html>

From imfanee at gmail.com  Thu Apr 30 10:40:50 2020
From: imfanee at gmail.com (Faisal Hanif)
Date: Thu, 30 Apr 2020 15:40:50 +0500
Subject: [Freeswitch-users] Calls per gateway
In-Reply-To: <002d01d61ed9$286f2f00$794d8d00$@delagarda.com>
References: <002d01d61ed9$286f2f00$794d8d00$@delagarda.com>
Message-ID: <CAO5BruCsqt8-HhnQr-UxwkrcSL6bNJp2Pxe6v_rfbOxydUwNJQ@mail.gmail.com>

mod_distributor could handle all this.

On Thu, 30 Apr 2020 at 15:26, Francesco Facco de Lagarda <
francesco at delagarda.com> wrote:

> Hello everyone .. I’ve been dredging the ‘net for a week trying to find a
> solution but can’t find one.
>
>
>
> I have a complicated system with 3 2n gateways with sim cards.
>
> Two have been dedicated to national calls and the last to international
> calls (the SIMS have been purchased with special deals according to the
> destination)
>
> I just need a simple method to find out how many current calls are on each
> gateway so I can route my calls accordingly. I use ESL to route my calls.
>
> I’ve tried all the variations of sofia status, show calls, show channels,
> but I cant find any that just tell me how many calls each gateway is
> handling at present (in and out)
>
>
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Regards,

Faisal Hanif
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200430/2a95d1f1/attachment.html>

From avi at avimarcus.net  Thu Apr 30 10:50:28 2020
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 30 Apr 2020 10:50:28 +0000
Subject: [Freeswitch-users] Calls per gateway
In-Reply-To: <002d01d61ed9$286f2f00$794d8d00$@delagarda.com>
References: <002d01d61ed9$286f2f00$794d8d00$@delagarda.com>
Message-ID: <01000171cab5903b-d0a0f509-fc12-4ff8-b6de-646e56434cdd-000000@email.amazonses.com>

Usually this is done with mod_limit, but I'm not sure how that will
interact with your ESL usage.

You may want to use it simply to track active calls.


On Thu, Apr 30, 2020, 1:21 PM Francesco Facco de Lagarda <
francesco at delagarda.com> wrote:

> Hello everyone .. I’ve been dredging the ‘net for a week trying to find a
> solution but can’t find one.
>
>
>
> I have a complicated system with 3 2n gateways with sim cards.
>
> Two have been dedicated to national calls and the last to international
> calls (the SIMS have been purchased with special deals according to the
> destination)
>
> I just need a simple method to find out how many current calls are on each
> gateway so I can route my calls accordingly. I use ESL to route my calls.
>
> I’ve tried all the variations of sofia status, show calls, show channels,
> but I cant find any that just tell me how many calls each gateway is
> handling at present (in and out)
>
>
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200430/ed8e5d24/attachment-0001.html>

From avi at avimarcus.net  Thu Apr 30 11:26:10 2020
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 30 Apr 2020 11:26:10 +0000
Subject: [Freeswitch-users] Calls per gateway
In-Reply-To: <CAO5BruCsqt8-HhnQr-UxwkrcSL6bNJp2Pxe6v_rfbOxydUwNJQ@mail.gmail.com>
References: <002d01d61ed9$286f2f00$794d8d00$@delagarda.com>
 <CAO5BruCsqt8-HhnQr-UxwkrcSL6bNJp2Pxe6v_rfbOxydUwNJQ@mail.gmail.com>
Message-ID: <01000171cad63e9a-5ece594a-9727-4376-ae8b-dd6eb6ba6aa9-000000@email.amazonses.com>

That looks like it's just balancing, how would you do limits with that?

On Thu, Apr 30, 2020, 2:21 PM Faisal Hanif <imfanee at gmail.com> wrote:

> mod_distributor could handle all this.
>
> On Thu, 30 Apr 2020 at 15:26, Francesco Facco de Lagarda <
> francesco at delagarda.com> wrote:
>
>> Hello everyone .. I’ve been dredging the ‘net for a week trying to find a
>> solution but can’t find one.
>>
>>
>>
>> I have a complicated system with 3 2n gateways with sim cards.
>>
>> Two have been dedicated to national calls and the last to international
>> calls (the SIMS have been purchased with special deals according to the
>> destination)
>>
>> I just need a simple method to find out how many current calls are on
>> each gateway so I can route my calls accordingly. I use ESL to route my
>> calls.
>>
>> I’ve tried all the variations of sofia status, show calls, show channels,
>> but I cant find any that just tell me how many calls each gateway is
>> handling at present (in and out)
>>
>>
>>
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
> Regards,
>
> Faisal Hanif
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200430/24d86808/attachment.html>

From francesco at delagarda.com  Thu Apr 30 12:30:52 2020
From: francesco at delagarda.com (Francesco Facco de Lagarda)
Date: Thu, 30 Apr 2020 14:30:52 +0200
Subject: [Freeswitch-users] Calls per gateway
In-Reply-To: <01000171cad63e9a-5ece594a-9727-4376-ae8b-dd6eb6ba6aa9-000000@email.amazonses.com>
References: <002d01d61ed9$286f2f00$794d8d00$@delagarda.com>
 <CAO5BruCsqt8-HhnQr-UxwkrcSL6bNJp2Pxe6v_rfbOxydUwNJQ@mail.gmail.com>
 <01000171cad63e9a-5ece594a-9727-4376-ae8b-dd6eb6ba6aa9-000000@email.amazonses.com>
Message-ID: <008c01d61eeb$31692310$943b6930$@delagarda.com>

Hy AVI

 

I have a complicated routing structure, and I do that all from db, managed by frontend.

I have a background process monitoring ESL, and it’s easy to monitor usage for outgoing calls, I can just use events “CHANNEL_CREATE” and “CHANNEL_HANGUP” and look at the “variable_sip_gateway_name”

But, with all the tons of info you get for every event, NEVER is the gateway name specified for incoming calls .. so I have no way to monitor usage on incoming.

I cant believe no fs_cli command exists to just check how many active calls you have on each gateway. Just for monitoring, this is an essential value, or am I missing something?

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Avi Marcus
Sent: giovedì 30 aprile 2020 13:26
To: FreeSWITCH Users Help <FreeSWITCH-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Calls per gateway

 

That looks like it's just balancing, how would you do limits with that? 

 

On Thu, Apr 30, 2020, 2:21 PM Faisal Hanif <imfanee at gmail.com <mailto:imfanee at gmail.com> > wrote:

mod_distributor could handle all this.

 

On Thu, 30 Apr 2020 at 15:26, Francesco Facco de Lagarda <francesco at delagarda.com <mailto:francesco at delagarda.com> > wrote:

Hello everyone .. I’ve been dredging the ‘net for a week trying to find a solution but can’t find one.

 

I have a complicated system with 3 2n gateways with sim cards.

Two have been dedicated to national calls and the last to international calls (the SIMS have been purchased with special deals according to the destination)

I just need a simple method to find out how many current calls are on each gateway so I can route my calls accordingly. I use ESL to route my calls.

I’ve tried all the variations of sofia status, show calls, show channels, but I cant find any that just tell me how many calls each gateway is handling at present (in and out)

 

 

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

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

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

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




 

-- 

Regards,

Faisal Hanif

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

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

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

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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200430/d5172dc1/attachment.html>

From asilva at wirelessmundi.com  Thu Apr 30 13:30:43 2020
From: asilva at wirelessmundi.com (=?utf-8?Q?Ant=C3=B3nio_Silva?=)
Date: Thu, 30 Apr 2020 15:30:43 +0200
Subject: [Freeswitch-users] mod_verto: subscribe events for specific
 accounts
In-Reply-To: <CAC7ttFF1z5DJcpUoTGTowZ4cmvxkW1PQw+Tbht3uVk2CaXae2g@mail.gmail.com>
References: <4DC90D2D-6FF9-4985-99F0-48A0565BFEDD@wirelessmundi.com>
 <mailman.2724.1588154883.1226.freeswitch-users@lists.freeswitch.org>
 <CAC7ttFF1z5DJcpUoTGTowZ4cmvxkW1PQw+Tbht3uVk2CaXae2g@mail.gmail.com>
Message-ID: <7D636856-0099-4154-9164-7982A99E45A0@wirelessmundi.com>

No problem. 

 Is “presence”   ( presene is an error...) 

> On 29 Apr 2020, at 12:23, Gregor Nanger <gregor at infomedia.si> wrote:
> 
> Thanx for sharing. Could be usefull.
> 
> One question. Is this right syntax:
> vertoHandle.subscribe(“presene”, { 
> 
> Is it presene or presence?
> 
> Best regards, Gregor 
> 
> On Wed, 29 Apr 2020 at 12:08, António Silva via FreeSWITCH-users <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>> wrote:
> 
> 
> 
> ---------- Forwarded message ----------
> From: "António Silva" <asilva at wirelessmundi.com <mailto:asilva at wirelessmundi.com>>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> Cc: 
> Bcc: 
> Date: Wed, 29 Apr 2020 12:07:29 +0200
> Subject: Re: mod_verto: subscribe events for specific accounts
> Reppling to myself, just found the right syntax,  it is posible from verto.js:
> 
> To get all events:
> vertoHandle.subscribe(“presene”, {
>                         handler: function(v, e) {
>                                 log.debug("***presence event e:", e);
>                         }
>                 });
> 
> But to get specific events per account:
> vertoHandle.subscribe([“presene.ACCOUNT1”, “presene.ACCOUNT2”], {
>                         handler: function(v, e) {
>                                 log.debug("***presence event e:", e);
>                         }
>                 });
> 
> Hope it help others :)
> 
> 
> > On 17 Mar 2020, at 13:53, António Silva <asilva at wirelessmundi.com <mailto:asilva at wirelessmundi.com>> wrote:
> > 
> > Hi,
> > 
> > Is it possible to subscribe using mod_verto?
> > 
> > Right now I’ve enable in mod_verto the presence events but I get all the events for all the accounts, and is fine but i was wondering if it’s is possible to just do a single subscribe like in sip.
> > 
> > Thanks for the help.
> > 
> > 
> > Regards,
> > António
> 
> 
> 
> 
> 
> ---------- Forwarded message ----------
> From: "António Silva via FreeSWITCH-users" <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> Cc: 
> Bcc: 
> Date: Wed, 29 Apr 2020 03:08:12 -0700 (PDT)
> Subject: Re: [Freeswitch-users] mod_verto: subscribe events for specific accounts
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/>
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community <https://signalwire.community/>
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com <mailto:sales at freeswitch.com>
> https://freeswitch.com <https://freeswitch.com/>
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss <https://freeswitch.com/oss>
> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
> https://cluecon.com <https://cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> https://freeswitch.com <https://freeswitch.com/>
> 
> -- 
> Gregor Nanger
>  
> CTO
> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 
> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia 
> • www.infomedia.si <http://www.infomedia.si/>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200430/ba78f689/attachment-0001.html>

From avi at avimarcus.net  Thu Apr 30 13:53:59 2020
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 30 Apr 2020 13:53:59 +0000
Subject: [Freeswitch-users] Calls per gateway
In-Reply-To: <008c01d61eeb$31692310$943b6930$@delagarda.com>
References: <002d01d61ed9$286f2f00$794d8d00$@delagarda.com>
 <CAO5BruCsqt8-HhnQr-UxwkrcSL6bNJp2Pxe6v_rfbOxydUwNJQ@mail.gmail.com>
 <01000171cad63e9a-5ece594a-9727-4376-ae8b-dd6eb6ba6aa9-000000@email.amazonses.com>
 <008c01d61eeb$31692310$943b6930$@delagarda.com>
Message-ID: <01000171cb5d9334-2acb442c-bc95-419a-9e86-d425e016cb24-000000@email.amazonses.com>

Oh, your issue is about incoming calls on those gateways?

If it's actually via the FreeSWITCH system of a gateway (instead of just an
open IP:PORT) you can have it set a variable that you can monitor for:

https://freeswitch.org/confluence/display/FREESWITCH/Gateways+Configuration#GatewaysConfiguration-Variables

<gateway>
 ...params...
 <variables>
*   <variable name="inbound_var_name"  value="this is inbound"
 direction="inbound"/>*
   <variable name="outbound_var_name" value="this is outbound"
direction="outbound"/>
   <variable name="both_var_name"     value="this on any direction"/>
 </variables>
</gateway>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200430/6a819101/attachment.html>

From davidswalkabout at gmail.com  Thu Apr 30 14:41:46 2020
From: davidswalkabout at gmail.com (David P)
Date: Thu, 30 Apr 2020 07:41:46 -0700
Subject: [Freeswitch-users] ICE TCP to reduce TURN cost
In-Reply-To: <mailman.2746.1588237959.1226.freeswitch-users@lists.freeswitch.org>
References: <mailman.2746.1588237959.1226.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAE23trtiMmv42Mk8=FQUs9q7Z8RwpVLrtu_snXPShyubkLCVFQ@mail.gmail.com>

Hey Brian,

I think I muddied the waters by referring to "TURN TCP" when I meant to be
asking about "ICE TCP"

Can FS be configured to support RFC 6544 ?
https://tools.ietf.org/html/rfc6544

Cheers,
David
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200430/391b7e83/attachment.html>

From aidar.kamalov at gmail.com  Tue Apr 28 09:44:21 2020
From: aidar.kamalov at gmail.com (Aidar Kamalov)
Date: Tue, 28 Apr 2020 12:44:21 +0300
Subject: [Freeswitch-users] freeswitch video conference recording black
	screen
In-Reply-To: <CAJs+-5rroJkHTUQnhvVh7qH8zNeD5sXR+9EM4=_zNGa=yTsXZA@mail.gmail.com>
References: <CAPXdxKh3DuZq47-suVZYCe-QTooAEsE2iz-6V9iy_HLZ2oKtLA@mail.gmail.com>
 <CAJs+-5rroJkHTUQnhvVh7qH8zNeD5sXR+9EM4=_zNGa=yTsXZA@mail.gmail.com>
Message-ID: <CAPXdxKi89JY=KO2v26f0rcPZPo3VBtGQ8_xhK_OV5JXhi4f-sg@mail.gmail.com>

fixed with <param name="video-mode" value="mux"/>

вт, 28 апр. 2020 г. в 12:20, Seven Du <dujinfang at gmail.com>:

> Did the first caller into the conference have video?
>
> On Wed, Apr 22, 2020 at 11:42 AM Aidar Kamalov <aidar.kamalov at gmail.com>
> wrote:
>
>> Hello, I am trying to record video conference:
>> <param name="auto-record"
>> value="$${recordings_dir}/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.mp4"/>
>>
>> But I am getting black screen with audio only in recording file. Although
>> I can record video on every single channels:
>> uuid_record UUID start /var/lib/freeswitch/recordings/UUID.mp4
>>
>> --
>> Aydar A. Kamalov
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Aydar A. Kamalov
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200428/79c8a809/attachment.html>

From Jazmin.Marino at on24.com  Thu Apr 30 01:24:57 2020
From: Jazmin.Marino at on24.com (Jazmin Marina Florez Marino)
Date: Thu, 30 Apr 2020 01:24:57 +0000
Subject: [Freeswitch-users] ADD 2 ICE CANDIDATES mod_verto
In-Reply-To: <CAEJMVkAvu1zP_Tpjm1d9uqfrcyKzp=QJuF203XhtE1XxThzF7g@mail.gmail.com>
References: <EAA0402198B6144FA20A2B37EB4E1A656E01E3E2@R-ESXMBX01.on24.com>
 <CAEJMVkAvu1zP_Tpjm1d9uqfrcyKzp=QJuF203XhtE1XxThzF7g@mail.gmail.com>
Message-ID: <EAA0402198B6144FA20A2B37EB4E1A656E01ED43@R-ESXMBX01.on24.com>

Hello Brian

I included that variable in vars.xml configuration file and after that change I get this.

CASE 1: When I added the ext-rtp-ip value in  verto.conf.xml, I got 2 ice candidates. However, they are the same (public ip). <param name="ext-rtp-ip" value="190.84.118.88"/>
CASE 2: When I commented the ext-rtp-ip value in verto.conf.xml  file, I only got local IP as ice candidate

In these configurations ext-rtp-ip in internal and external sip_profiles are set with auto-nat values.

How can I get both IPs (local and public) as valid Ice candidates in the SDP?

CASE 1
2020-04-29 19:50:56.535350 [DEBUG] switch_core_media.c:8498 Audio params are unchanged for verto.rtc/5715976.
2020-04-29 19:50:56.535350 [DEBUG] mod_verto.c:2502 Local SDP verto.rtc/5715976:
v=0
o=FreeSWITCH 1588188372 1588188373 IN IP4 190.84.118.88
s=FreeSWITCH
c=IN IP4 190.84.118.88
t=0 0
a=msid-semantic: WMS RQn2gdt1uxyuNrfyPxHSSO3S0hq5M1jt
m=audio 19484 UDP/TLS/RTP/SAVPF 111 110
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1; minptime=10; stereo=1; sprop-stereo=1
a=rtpmap:110 telephone-event/48000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=fingerprint:sha-256 FD:DF:48:02:C3:F8:D9:57:F8:C8:97:CD:3A:AB:6A:A6:97:4A:BA:A5:48:5A:3A:2D:0F:18:14:EC:D5:A9:21:86
a=setup:active
a=rtcp-mux
a=rtcp:19484 IN IP4 190.84.118.88
a=ice-ufrag:ICIqgB97Jp2mgcFO
a=ice-pwd:yRsZnXKhQPN10ISzbqQhSPes
a=candidate:2978044633 1 udp 659136 190.84.118.88 19484 typ host generation 0
a=candidate:3048048785 1 udp 659136 190.84.118.88 19484 typ host generation 0
a=end-of-candidates
a=ssrc:917829176 cname:HwmNAFFU33fGlk40
a=ssrc:917829176 msid:RQn2gdt1uxyuNrfyPxHSSO3S0hq5M1jt a0
a=ssrc:917829176 mslabel:RQn2gdt1uxyuNrfyPxHSSO3S0hq5M1jt
a=ssrc:917829176 label:RQn2gdt1uxyuNrfyPxHSSO3S0hq5M1jta0
m=video 0 UDP/TLS/RTP/SAVPF 19


CASE 2
2020-04-29 20:05:01.563371 [DEBUG] mod_verto.c:2502 Local SDP verto.rtc/5715976:
v=0
o=FreeSWITCH 1588191001 1588191002 IN IP4 192.168.1.170
s=FreeSWITCH
c=IN IP4 192.168.1.170
t=0 0
a=msid-semantic: WMS uwE5jFb2y1vHfnFvS07p6MR0MiZh3hfo
m=audio 17700 UDP/TLS/RTP/SAVPF 111 110
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1; minptime=10; stereo=1; sprop-stereo=1
a=rtpmap:110 telephone-event/48000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=fingerprint:sha-256 FD:DF:48:02:C3:F8:D9:57:F8:C8:97:CD:3A:AB:6A:A6:97:4A:BA:A5:48:5A:3A:2D:0F:18:14:EC:D5:A9:21:86
a=setup:active
a=rtcp-mux
a=rtcp:17700 IN IP4 192.168.1.170
a=ice-ufrag:yBbvdTlQ2aXVi29M
a=ice-pwd:CipItVdvMfCRN6EcxGxApIhh
a=candidate:8338699166 1 udp 659136 192.168.1.170 17700 typ host generation 0
a=end-of-candidates
a=ssrc:1119156645 cname:EKjNgkFEb1cheIDW
a=ssrc:1119156645 msid:uwE5jFb2y1vHfnFvS07p6MR0MiZh3hfo a0
a=ssrc:1119156645 mslabel:uwE5jFb2y1vHfnFvS07p6MR0MiZh3hfo
a=ssrc:1119156645 label:uwE5jFb2y1vHfnFvS07p6MR0MiZh3hfoa0
m=video 0 UDP/TLS/RTP/SAVPF 19




De: Brian West <brian at freeswitch.com>
Enviado el: miércoles, 29 de abril de 2020 15:17
Para: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Asunto: Re: [Freeswitch-users] ADD 2 ICE CANDIDATES mod_verto

set the variable include_external_ip=true

/b

On Mon, Apr 27, 2020 at 3:02 PM Jazmin Marina Florez Marino <Jazmin.Marino at on24.com<mailto:Jazmin.Marino at on24.com>> wrote:
Hello all,

I am working with mod_verto in a scenario where I need to send public and private IP addresses as ICE candidates.
When I modify the ext-rtp-ip parameter, it only sends one option as typ host ice candidate. However, for my implementation I need to send both.

What I need is the combination of those ice candidates, something like this:
a=candidate:8228103366 1 udp 659136 190.84.118.88 31028 typ host generation 0
a=candidate:3496438569 1 udp 659136 192.168.1.170 24210 typ host generation 0

Is it possible to add both ice candidates in the same SDP from configuration files?


SDP with local IP
2020-04-24 14:17:23.715620 [DEBUG] mod_verto.c:2502 Local SDP verto.rtc/5715976:
v=0
o=FreeSWITCH 1587724815 1587724816 IN IP4 190.84.118.88
s=FreeSWITCH
c=IN IP4 190.84.118.88
t=0 0
a=msid-semantic: WMS 8gumHFWDC0sD12iPvVhSeEhUfLdXQBVO
m=audio 31028 UDP/TLS/RTP/SAVPF 111 110
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1; minptime=10; stereo=1; sprop-stereo=1
a=rtpmap:110 telephone-event/48000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=fingerprint:sha-256 FD:DF:48:02:C3:F8:D9:57:F8:C8:97:CD:3A:AB:6A:A6:97:4A:BA:A5:48:5A:3A:2D:0F:18:14:EC:D5:A9:21:86
a=setup:active
a=rtcp-mux
a=rtcp:31028 IN IP4 190.84.118.88
a=ice-ufrag:BiYFM6pSItFoUq2t
a=ice-pwd:wTz7x9b00ImLonRPvGl8CBRb
a=candidate:8228103366 1 udp 659136 190.84.118.88 31028 typ host generation 0
a=end-of-candidates
a=ssrc:2930644107 cname:If1gLNJdD1HvUPbg
a=ssrc:2930644107 msid:8gumHFWDC0sD12iPvVhSeEhUfLdXQBVO a0
a=ssrc:2930644107 mslabel:8gumHFWDC0sD12iPvVhSeEhUfLdXQBVO
a=ssrc:2930644107 label:8gumHFWDC0sD12iPvVhSeEhUfLdXQBVOa0
m=video 0 UDP/TLS/RTP/SAVPF 19

SDP with Public IP
<param name="ext-rtp-ip" value="190.84.118.88"/>
2020-04-24 14:13:49.395900 [DEBUG] mod_verto.c:2502 Local SDP verto.rtc/5715976:
v=0
o=FreeSWITCH 1587731419 1587731420 IN IP4 192.168.1.170
s=FreeSWITCH
c=IN IP4 192.168.1.170
t=0 0
a=msid-semantic: WMS t9mERteVPrwT3aZbexPlp2U6Cspg78dq
m=audio 24210 UDP/TLS/RTP/SAVPF 111 110
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1; minptime=10; stereo=1; sprop-stereo=1
a=rtpmap:110 telephone-event/48000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=fingerprint:sha-256 FD:DF:48:02:C3:F8:D9:57:F8:C8:97:CD:3A:AB:6A:A6:97:4A:BA:A5:48:5A:3A:2D:0F:18:14:EC:D5:A9:21:86
a=setup:active
a=rtcp-mux
a=rtcp:24210 IN IP4 192.168.1.170
a=ice-ufrag:mScCKXydfOjcMntK
a=ice-pwd:Depc4PrcDqdqfgirZlTz8iCe
a=candidate:3496438569 1 udp 659136 192.168.1.170 24210 typ host generation 0
a=end-of-candidates
a=ssrc:3132020181 cname:7pVPJ0AphX8bXYd1
a=ssrc:3132020181 msid:t9mERteVPrwT3aZbexPlp2U6Cspg78dq a0
a=ssrc:3132020181 mslabel:t9mERteVPrwT3aZbexPlp2U6Cspg78dq
a=ssrc:3132020181 label:t9mERteVPrwT3aZbexPlp2U6Cspg78dqa0
m=video 0 UDP/TLS/RTP/SAVPF 19

2020-04-24 14:13:49.395900 [NOTICE] mod_dptools.c:1360 Channel [verto.rtc/5715976] has been answered

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

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

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

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


--


Brian West | Co-founder and Developer

Need Commercial support? email sales at freeswitch.com<mailto:sales at freeswitch.com>

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

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

Mobile: 918-424-9378

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

[https://www.facebook.com/signalwireinc?src=email]<https://www.facebook.com/freeswitch>[https://twitter.com/freeswitch]<https://twitter.com/freeswitch>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200430/6a87845a/attachment-0001.html>

From lafuente_sonny at yahoo.com  Tue Apr 28 12:32:19 2020
From: lafuente_sonny at yahoo.com (Sonny Lafuente)
Date: Tue, 28 Apr 2020 20:32:19 +0800
Subject: Auto Changing RTP concern
References: <1C925022-50D8-4096-AD1D-A33AB1A259C5.ref@yahoo.com>
Message-ID: <1C925022-50D8-4096-AD1D-A33AB1A259C5@yahoo.com>

Hello Everyone,

Does anyone here experiencing the “Auto changing RTP port” in freeswitch? It actually happened when initiating a call from local SIP going to SIP server installed in the CLOUD. There is no Audio both ends after the SIP call setup. Now, when initiating call from cloud going to local SIP I have no experience in Intermittent RTP. Btw, the device that is registered to the SIP server in the cloud is Polycom IP Phone. 

Basically, I have SIP server in cloud and a SIP server in local. The local SIP server is using VPN to tunnel the RTP because it is behind NAT. Instead of doing port forwarding.

I have tried the following:

1. In the freeswitch, I configured the ”auto_adjust_rtp=true”

2. in the local freeswitch, I configured the internal and external profile to ext-rtp-ip=<VPN IP>

If anyone who is experiencing this and able to solve the issue, please give me some point on what I need to check. 

Thank you.

SL

Sent from my iPhone


From m.hald at outlook.com  Tue Apr 28 11:22:01 2020
From: m.hald at outlook.com (Marcel Haldemann)
Date: Tue, 28 Apr 2020 11:22:01 +0000
Subject: [Freeswitch-users] curl zombie_exec
In-Reply-To: <CAJs+-5rHr=O79AJUMVTEHGoxBUF12jPtDffKQ1Kks8=bUTgQ=Q@mail.gmail.com>
References: <FABBEDAF-2213-49C4-89A4-F4E8FC53D447@telepro.me>
 <CAJs+-5rHr=O79AJUMVTEHGoxBUF12jPtDffKQ1Kks8=bUTgQ=Q@mail.gmail.com>
Message-ID: <PR3PR01MB6666958F2E8AF2F55C711CF29EAC0@PR3PR01MB6666.eurprd01.prod.exchangelabs.com>

Would be a nice patch.
Currently I’m using an ugly hack, doing it like so (as os.execute seems to have the zombie flag):

        dat = env:serialize("json")
        res = os.execute("curl --header \"Content-Type: application/json\" --connect-timeout 30 --max-time 30 --request POST --data '" .. dat .. "' http://127.0.0.1:8090/api/dialplan > /dev/null 2>&1 &")


Von: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> Im Auftrag von Seven Du
Gesendet: Dienstag, 28. April 2020 10:22
An: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Betreff: Re: [Freeswitch-users] curl zombie_exec

you have to patch mod_curl do to that like below. please make a pr if it works for you.

    SWITCH_ADD_APP(app_interface, "curl", "Perform a http request", "Perform a http request",
                   curl_app_function, SYNTAX, SAF_SUPPORT_NOMEDIA | SAF_ROUTING_EXEC | SAF_ZOMBIE_EXEC);


On Tue, Apr 28, 2020 at 4:07 AM Mai Duy Quang <quang at telepro.me<mailto:quang at telepro.me>> wrote:
Hi,

I call curl in lua script but has this error, been googling all day long but no hope. Any have fix for that?

Channel is hungup and application curl does not have the zombie_exec flag.

my lua code:
-- to allow curl after the call
session:execute("set_zombie_exec");

freeswitch.consoleLog("notice", "[curl][ring groups][missed]  " .. missed_call_data .. "  ");

session:execute("curl", missed_call_data .. " phone=" .. caller_id_number .. "&target=" .. ring_group_name);
curl_response_code = session:getVariable("curl_response_code")
curl_response      = session:getVariable("curl_response_data”)


Many thanks,
Johnathan Mai
_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com<https://nam10.safelinks.protection.outlook.com/?url=https%3A%2F%2Fsignalwire.com%2F&data=02%7C01%7C%7C16b369bad9e2498c146508d7eb50ec91%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C637236605101588615&sdata=LMETlyBxNIf%2B3dEOydL5plCK0zCVqfZ%2FEPOFdf6QnW8%3D&reserved=0>
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community<https://nam10.safelinks.protection.outlook.com/?url=https%3A%2F%2Fsignalwire.community%2F&data=02%7C01%7C%7C16b369bad9e2498c146508d7eb50ec91%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C637236605101588615&sdata=J31UDdSv6kkEKeCG9NKJPs9YR6XMw22YPQV1AFjfEKY%3D&reserved=0>

Professional FreeSWITCH Services
sales at freeswitch.com<mailto:sales at freeswitch.com>
https://freeswitch.com<https://nam10.safelinks.protection.outlook.com/?url=https%3A%2F%2Ffreeswitch.com%2F&data=02%7C01%7C%7C16b369bad9e2498c146508d7eb50ec91%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C637236605101598611&sdata=ZJNjUvLMMfeVH3M9w2MRlV41qJQpn3Mmg5HMMTyVxPo%3D&reserved=0>

Official FreeSWITCH Sites
https://freeswitch.com/oss<https://nam10.safelinks.protection.outlook.com/?url=https%3A%2F%2Ffreeswitch.com%2Foss&data=02%7C01%7C%7C16b369bad9e2498c146508d7eb50ec91%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C637236605101598611&sdata=RWSm5I47FgDOyuvAvjgDDfnJZkTi3XPDWs79pSHYYnQ%3D&reserved=0>
https://freeswitch.org/confluence<https://nam10.safelinks.protection.outlook.com/?url=https%3A%2F%2Ffreeswitch.org%2Fconfluence&data=02%7C01%7C%7C16b369bad9e2498c146508d7eb50ec91%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C637236605101598611&sdata=333wxW42hnkrkiTsQ0sGWIyy2L014bdvdIUG6eq%2F8p4%3D&reserved=0>
https://cluecon.com<https://nam10.safelinks.protection.outlook.com/?url=https%3A%2F%2Fcluecon.com%2F&data=02%7C01%7C%7C16b369bad9e2498c146508d7eb50ec91%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C637236605101608602&sdata=6jcfxdseSyptWPHMVSwrJVVGyrjy5hCYqqnIo%2BDTr%2FM%3D&reserved=0>

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<https://nam10.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7C%7C16b369bad9e2498c146508d7eb50ec91%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C637236605101608602&sdata=2pnT8BU6NBMW82pV0g51dwdEG2rHy%2Bu21ckRel7vdZs%3D&reserved=0>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<https://nam10.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Foptions%2Ffreeswitch-users&data=02%7C01%7C%7C16b369bad9e2498c146508d7eb50ec91%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C637236605101618599&sdata=JJ%2BsngM%2BX%2FYoDLX2LwNw1TYooJucMDrkUKWPCALisb0%3D&reserved=0>
https://freeswitch.com<https://nam10.safelinks.protection.outlook.com/?url=https%3A%2F%2Ffreeswitch.com%2F&data=02%7C01%7C%7C16b369bad9e2498c146508d7eb50ec91%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C637236605101618599&sdata=Afbk%2FtuEKY9FnWr0KnRmYywxSlljEBTeaAC%2F24ntBl0%3D&reserved=0>


--
About: http://about.me/dujinfang<https://nam10.safelinks.protection.outlook.com/?url=http%3A%2F%2Fabout.me%2Fdujinfang&data=02%7C01%7C%7C16b369bad9e2498c146508d7eb50ec91%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C637236605101618599&sdata=Xqq5fkV0IZ24kgMg3doSKrI0GaAQzam1S0%2FU2Ibaqts%3D&reserved=0>
Blog: http://www.dujinfang.com<https://nam10.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.dujinfang.com%2F&data=02%7C01%7C%7C16b369bad9e2498c146508d7eb50ec91%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C637236605101628595&sdata=KwNCzQCE3gSBo56%2Bft3aMmOaewUq9STaT%2BMas54h%2Byg%3D&reserved=0>
Proj:  http://www.freeswitch.org.cn<https://nam10.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org.cn%2F&data=02%7C01%7C%7C16b369bad9e2498c146508d7eb50ec91%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C637236605101628595&sdata=7bG6yXsmBxeU5godYW4mgM0J6IWQ5bqpLYwZELcRTog%3D&reserved=0>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200428/db915106/attachment-0001.html>

From moshe.rosenberg at gmail.com  Tue Apr 28 12:28:29 2020
From: moshe.rosenberg at gmail.com (Moshe Rosenberg)
Date: Tue, 28 Apr 2020 08:28:29 -0400
Subject: [Freeswitch-users] Announcing SaraPhone,
 SIP WebRTC Open Source business phone
In-Reply-To: <CALXCt0o5f758qjP24A9y1z_KAKOGcUz9kaTsyA8fo4k+2tt8nw@mail.gmail.com>
References: <CALXCt0qaDgo0gSmkKSy0uzK+5v3tS6KRJgXN78u2qFWL6JH4MA@mail.gmail.com>
 <CAAoGpGSzph5NMAmDNzLgvQrG2_w7zLVi8UG7UXYnf6M2O_sPqQ@mail.gmail.com>
 <CAJs+-5ovMbpnLz7g4Erivh_QdrFOqzZEZNy8euCAha14sTZcKQ@mail.gmail.com>
 <CALXCt0o5f758qjP24A9y1z_KAKOGcUz9kaTsyA8fo4k+2tt8nw@mail.gmail.com>
Message-ID: <CAL1=tOTSQsD+jpc9CTh1=WWJVjF09=+KLB9N8WyEbwTiC-rxhg@mail.gmail.com>

https://github.com/gmaruzz/saraphone/blob/master/doc/INSTALL.txt

Why are you using preset passwords for fusion and the DB.

Why not random like the default script

On Tue, Apr 28, 2020, 5:43 AM Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:

> Seven <3
>
>
>
> On Tue, Apr 28, 2020 at 11:17 AM Seven Du <dujinfang at gmail.com> wrote:
>
>> Nice ;)
>>
>> On Fri, Apr 17, 2020 at 2:31 AM Mitul Limbani <mitul at enterux.com> wrote:
>>
>>> Great Giovanni, shall definitely try this in our environment :-)
>>>
>>> Mitul Limbani
>>>
>>> On Thu, 16 Apr, 2020, 21:05 Giovanni Maruzzelli, <gmaruzz at gmail.com>
>>> wrote:
>>>
>>>> My fellow VoIPers,
>>>>
>>>> I am pleased to announce the early availability of:
>>>>
>>>> SaraPhone
>>>> ------------------
>>>>
>>>> SaraPhone is a bare bone SIP WebRTC voice phone, complete with most
>>>> features real companies want to use in real world: HotDesking, Redial,
>>>> BLFs, MWI, DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer,
>>>> Notifications, running on all Browsers both on Desktop and SmartPhone.
>>>>
>>>> SaraPhone is fully integrated with FusionPBX, the full-featured domain
>>>> based multi-tenant PBX and voice switch for FreeSwitch.
>>>>
>>>> Based on SIP.js, SaraPhone works with all WebRTC compliant SIP proxies,
>>>> gateways, and servers (Asterisk, OpenSIPS, Kamailio, etc).
>>>>
>>>> Initial author is Giovanni Maruzzelli, and SaraPhone gets its name from
>>>> Giovanni's wife, Sara Hosseini.
>>>>
>>>> In addition to providing all of the usual DeskPhone functionality,
>>>> SaraPhone got:
>>>>
>>>>    - Desktop Notification for Incoming Calls
>>>>    - Live MWI update
>>>>    - Real Time BLFs status update
>>>>    - BLF click to call
>>>>    - Caller Name and Number Display
>>>>    - Call Error Cause Display
>>>>    - AutoAnswer
>>>>    - Network Disconnect Reload
>>>>    - Show and Set Caller-ID (incoming-outbound)
>>>>
>>>>
>>>> You an find it in GitHub ( https://github.com/gmaruzz/saraphone ).
>>>>
>>>> Anyone interested can play with it :).
>>>>
>>>> Have fun,
>>>> giovanni
>>>>
>>>>
>>>> --
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> OpenTelecom.IT
>>>> cell: +39 347 266 56 18
>>>>
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>>
>>
>> --
>> About: http://about.me/dujinfang
>> Blog: http://www.dujinfang.com
>> Proj:  http://www.freeswitch.org.cn
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200428/b47511d4/attachment.html>

From whiskeykontequila at gmail.com  Wed Apr 29 06:13:46 2020
From: whiskeykontequila at gmail.com (Aleksandar Bilanovic)
Date: Wed, 29 Apr 2020 08:13:46 +0200
Subject: [Freeswitch-users] Need help with mod_v8 install
In-Reply-To: <CAPxXe8V+RZpHFvLSgfiWoqp1rp+muZPrjzP3wPNT8JYEiYb89g@mail.gmail.com>
References: <CAPxXe8V+RZpHFvLSgfiWoqp1rp+muZPrjzP3wPNT8JYEiYb89g@mail.gmail.com>
Message-ID: <CAP0BNaJtrHrZf_2Tqje=gcAB5_dDDOav0-cX5JnVXTwAiTsdxQ@mail.gmail.com>

You are missing v8 headers. What Linux platform are you using?

Aleksandar

On Mon, Apr 27, 2020 at 10:14 PM Pete Kay <petedao at gmail.com> wrote:

> Hi
>
>
> I am getting error when trying to install mod_v8.
>
>
> Could someone pls help?
>
>
> # make
>
> make  all-am
>
> make[1]: Entering directory `/usr/src/freeswitch/src/mod/languages/mod_v8'
>
>   CXX      mod_v8_la-mod_v8.lo
>
> In file included from *mod_v8.h:36:0*,
>
>                  from *mod_v8.cpp:68*:
>
> *./include/javascript.hpp:35:16:* *fatal error: *v8.h: No such file or
> directory
>
>  #include <v8.h>
>
> *                ^*
>
> compilation terminated.
>
> make[1]: *** [mod_v8_la-mod_v8.lo] Error 1
>
> make[1]: Leaving directory `/usr/src/freeswitch/src/mod/languages/mod_v8'
>
> make: *** [all] Error 2
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Aleksandar Bilanovic
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200429/a5a4de38/attachment-0001.html>