From happy.neko at gmail.com  Wed Jun  1 00:30:13 2016
From: happy.neko at gmail.com (happy.neko)
Date: Tue, 31 May 2016 23:30:13 +0300
Subject: [Freeswitch-users] Low cost voip phone ?
In-Reply-To: <jxhm3jmfynjdhvghugwcd774.1464709645365@email.android.com>
References: <xwblnhn6bbxbmii9dsid86wv.1464707474412@email.android.com>
	<CAHzfiZ7JkNMzWYGsm0A1vsyMmE0wLvj9rcRhPquCuPzfJKOTOA@mail.gmail.com>
	<5p4x48mh6rkjfx8v2f1smhrw.1464708465158@email.android.com>
	<CAH=YQ8skz9zzLnFzHLVMU9Cvt_GbgE3MWZRmj=zEQ6PMD4hXjA@mail.gmail.com>
	<jxhm3jmfynjdhvghugwcd774.1464709645365@email.android.com>
Message-ID: <CAH=YQ8vTpW3kPBjzgedKCAYBpmRQ65PpOB-aYAwpZTvqdDB97g@mail.gmail.com>

2016-05-31 18:47 GMT+03:00 Shawn Wheeler <
Shawn.Wheeler at interlockconcepts.com>:

>
> I have heard about the dial plan and found it in the gui but I have not
> been successful in finding what it needs to be set to.
>
> Dialplan sets how many digits device expects to be pressed before actually
placing the call. You may configure it to set local numbers to be exactly
10 digits etc.
As Bob said try to press SEND or # key immediately after the phone number
to see if it makes the difference.
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From Shawn.Wheeler at interlockconcepts.com  Wed Jun  1 00:45:46 2016
From: Shawn.Wheeler at interlockconcepts.com (Shawn Wheeler)
Date: Tue, 31 May 2016 20:45:46 +0000
Subject: [Freeswitch-users] Low cost voip phone ?
In-Reply-To: <CAH=YQ8vTpW3kPBjzgedKCAYBpmRQ65PpOB-aYAwpZTvqdDB97g@mail.gmail.com>
References: <xwblnhn6bbxbmii9dsid86wv.1464707474412@email.android.com>
	<CAHzfiZ7JkNMzWYGsm0A1vsyMmE0wLvj9rcRhPquCuPzfJKOTOA@mail.gmail.com>
	<5p4x48mh6rkjfx8v2f1smhrw.1464708465158@email.android.com>
	<CAH=YQ8skz9zzLnFzHLVMU9Cvt_GbgE3MWZRmj=zEQ6PMD4hXjA@mail.gmail.com>
	<jxhm3jmfynjdhvghugwcd774.1464709645365@email.android.com>,
	<CAH=YQ8vTpW3kPBjzgedKCAYBpmRQ65PpOB-aYAwpZTvqdDB97g@mail.gmail.com>
Message-ID: <p4acphb0apbwj94m3c8v992c.1464727543204@email.android.com>

It has the default setting in it at the moment.  And I am using 4 digits as this phone is for internal only.  I wl give send and or pound a try.  Thank you



Sent from my Verizon, Samsung Galaxy smartphone


-------- Original message --------
From: "happy.neko" <happy.neko at gmail.com>
Date: 5/31/16 14:33 (GMT-07:00)
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Low cost voip phone ?

2016-05-31 18:47 GMT+03:00 Shawn Wheeler <Shawn.Wheeler at interlockconcepts.com<mailto:Shawn.Wheeler at interlockconcepts.com>>:

I have heard about the dial plan and found it in the gui but I have not been successful in finding what it needs to be set to.

Dialplan sets how many digits device expects to be pressed before actually placing the call. You may configure it to set local numbers to be exactly 10 digits etc.
As Bob said try to press SEND or # key immediately after the phone number to see if it makes the difference.
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From jkomar at jbox.ca  Wed Jun  1 00:50:21 2016
From: jkomar at jbox.ca (Jason Komar)
Date: Tue, 31 May 2016 14:50:21 -0600
Subject: [Freeswitch-users] Low cost voip phone ?
In-Reply-To: <p4acphb0apbwj94m3c8v992c.1464727543204@email.android.com>
References: <xwblnhn6bbxbmii9dsid86wv.1464707474412@email.android.com>
	<CAHzfiZ7JkNMzWYGsm0A1vsyMmE0wLvj9rcRhPquCuPzfJKOTOA@mail.gmail.com>
	<5p4x48mh6rkjfx8v2f1smhrw.1464708465158@email.android.com>
	<CAH=YQ8skz9zzLnFzHLVMU9Cvt_GbgE3MWZRmj=zEQ6PMD4hXjA@mail.gmail.com>
	<jxhm3jmfynjdhvghugwcd774.1464709645365@email.android.com>
	<CAH=YQ8vTpW3kPBjzgedKCAYBpmRQ65PpOB-aYAwpZTvqdDB97g@mail.gmail.com>
	<p4acphb0apbwj94m3c8v992c.1464727543204@email.android.com>
Message-ID: <CA+zebiYKUJTOsow4uRTH3keeKJh=TGpTQh=e5M529=nMLPX6oA@mail.gmail.com>

If you are using only 4 numbers, try changing the dial plan in the phone to
{xxxx}

Jason


On Tue, May 31, 2016 at 2:45 PM, Shawn Wheeler <
Shawn.Wheeler at interlockconcepts.com> wrote:

> It has the default setting in it at the moment.  And I am using 4 digits
> as this phone is for internal only.  I wl give send and or pound a try.
> Thank you
>
>
>
> Sent from my Verizon, Samsung Galaxy smartphone
>
>
> -------- Original message --------
> From: "happy.neko" <happy.neko at gmail.com>
> Date: 5/31/16 14:33 (GMT-07:00)
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Low cost voip phone ?
>
> 2016-05-31 18:47 GMT+03:00 Shawn Wheeler <
> Shawn.Wheeler at interlockconcepts.com>:
>
>>
>> I have heard about the dial plan and found it in the gui but I have not
>> been successful in finding what it needs to be set to.
>>
>> Dialplan sets how many digits device expects to be pressed before
> actually placing the call. You may configure it to set local numbers to be
> exactly 10 digits etc.
> As Bob said try to press SEND or # key immediately after the phone number
> to see if it makes the difference.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From colin.morelli at gmail.com  Wed Jun  1 00:59:15 2016
From: colin.morelli at gmail.com (Colin Morelli)
Date: Tue, 31 May 2016 20:59:15 +0000
Subject: [Freeswitch-users] Bridge Call and Join Conference
Message-ID: <CAPtU-UqO_jVO1OFxthu7TkqWCitUpwcU2E3SU0p0QmpXJ09MMQ@mail.gmail.com>

So, I'm trying to setup certain calls such that they enter into a
conference asap. I have the process working using bridging conferences
right now, which is okay, but bridging conferences forces the dialing leg
of the call to go to an answered state immediately, while the bridge leg is
still ringing. So, while everything seems to work correctly, it messes with
some of my reconciliation/reporting process with another database that are
fed off of channel state events. Essentially it looks like every call was
answered.

So I was curious to know if there's a way to make a call that creates a
bridged leg, and then as soon as the leg is answered immediately joins both
legs into a conference. Is the only way to do it to listen for the
channel_bridge events and then transfer the call myself?

Best,
Colin
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From schoch+freeswitch.org at xwin32.com  Wed Jun  1 03:35:47 2016
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Tue, 31 May 2016 16:35:47 -0700
Subject: [Freeswitch-users] Run a LUA script when voicemail received
In-Reply-To: <CAOkuKde_1pK6TzA4-9qKSVuh8A0s7i1qs2n4uUt-5-b8=HSgKQ@mail.gmail.com>
References: <CAOkuKdfnwFsKN4H7ZxB+dvfGSGnAby59AM+kO4MCrLuJWa=Qag@mail.gmail.com>
	<CAEwTsAg0rPXMtvJtGns1kwcNWRT2rHJSnn+psXfnF9_BMdV3Ww@mail.gmail.com>
	<CAFmZFiNHye2Tujfwsw8mh9tc+bnZNxSbxTt670W0L3UtCKszGg@mail.gmail.com>
	<CAOkuKdczNFzL76oRF1f2SARJ7700owGmMnJOrtTZA6x=5AsLJQ@mail.gmail.com>
	<CAOkuKde_1pK6TzA4-9qKSVuh8A0s7i1qs2n4uUt-5-b8=HSgKQ@mail.gmail.com>
Message-ID: <CAOkuKdcvRDVkmOhdMQ+7VB8edhdbzMX+25-wRAxgPSn-waX4iw@mail.gmail.com>

I'm back to working on the voicemail notification.
The lua script is being called, and I can get important variables with
these calls:

action = event:getHeader("VM-Action")
if action ~= "leave-message" then return end
user = event:getHeader("VM-User")
caller_id_name = event:getHeader("VM-Caller-ID-Name")
caller_id_number = event:getHeader("VM-Caller-ID-Number")
uuid = event:getHeader("VM-UUID")
message_len = event:getHeader("VM-Message-Len")
timestamp = event:getHeader("VM-Timestamp")

I have some of the important information here, like which voicemail box got
the message, who's it from, when, and how long it was. At this point I'd
like to send this information as an SMS message using a CURL call into the
Flowroute SMS API. The important bit of data is to whom to send the SMS
message.

I put the SMS number into the directory XML structure, like this:
<include>
  <user id="1004">
    <params>
      <param name="vm-mailto" value="user at example.com"/>
    </params>
    <variables>
      <variable name="effective_caller_id_name" value="Joe Smith"/>
      <variable name="effective_caller_id_number" value="1004"/>
      <variable name="outbound_caller_id_name"
value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number"
value="$${outbound_caller_id}"/>
      <variable name="vm-notify-sms" value="1408497xxxx"/>
    </variables>
  </user>
</include>

I'd like to pull that vm-notify-sms variable that I created. The documented
way to get variables from a lua script is the session:getVariable()
function, but the session variable is not defined when this event hook is
called. However, the log file shows my script is being called before the
session is destroyed, so I'm thinking perhaps I can access it from the uuid
variable. How do I do this? I also looked for an "env" object, which is
created when a hangup hook is called, but it seems that is only set on a
hangup hook, not an event hook.

How can I access the session or directory variables?

-- 
Steve
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From schoch+freeswitch.org at xwin32.com  Wed Jun  1 04:10:04 2016
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Tue, 31 May 2016 17:10:04 -0700
Subject: [Freeswitch-users] Run a LUA script when voicemail received
In-Reply-To: <CAOkuKdcvRDVkmOhdMQ+7VB8edhdbzMX+25-wRAxgPSn-waX4iw@mail.gmail.com>
References: <CAOkuKdfnwFsKN4H7ZxB+dvfGSGnAby59AM+kO4MCrLuJWa=Qag@mail.gmail.com>
	<CAEwTsAg0rPXMtvJtGns1kwcNWRT2rHJSnn+psXfnF9_BMdV3Ww@mail.gmail.com>
	<CAFmZFiNHye2Tujfwsw8mh9tc+bnZNxSbxTt670W0L3UtCKszGg@mail.gmail.com>
	<CAOkuKdczNFzL76oRF1f2SARJ7700owGmMnJOrtTZA6x=5AsLJQ@mail.gmail.com>
	<CAOkuKde_1pK6TzA4-9qKSVuh8A0s7i1qs2n4uUt-5-b8=HSgKQ@mail.gmail.com>
	<CAOkuKdcvRDVkmOhdMQ+7VB8edhdbzMX+25-wRAxgPSn-waX4iw@mail.gmail.com>
Message-ID: <CAOkuKddVsmLA-+YjQo7myFF7HXta6K468t+7WooETYU6=_vJkw@mail.gmail.com>

I may have just answered my own question again. I found a way, but please
let me know if you know of a cleaner way:

user = event:getHeader("VM-User")
domain = event:getHeader("VM-Domain")
api = freeswitch.API();
from = api:executeString("user_data " ..user .. "@" .. domain ..  " var
outbound_caller_id_number")
to = api:executeString("user_data " .. user .. "@" .. domain .. " param
vm-notify-sms")

This should get me what I need to send the text.

-- 
Steve
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From dujinfang at gmail.com  Wed Jun  1 04:52:01 2016
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 1 Jun 2016 08:52:01 +0800
Subject: [Freeswitch-users] SIP Message question
In-Reply-To: <f8bf939768814dbaa7c2e2639e63eac0@r5prod-exchange.robot5.de>
References: <f8bf939768814dbaa7c2e2639e63eac0@r5prod-exchange.robot5.de>
Message-ID: <326EEA13-5AA6-45EE-965B-AD02F8E1C8CF@gmail.com>

it goes to chatplan

https://freeswitch.org/confluence/display/FREESWITCH/mod_sms <https://freeswitch.org/confluence/display/FREESWITCH/mod_sms>


> On May 30, 2016, at 8:27 PM, Nagorny, Dimitry <dimitry.nagorny at robot5.de> wrote:
> 
> Hello List,
>  
> AFAIK, all INVITE SIP messages (INVITE sip:100 at mydomain.com <sip:100 at mydomain.com>) are going through the set up dialplans. Are all MESSAGE SIP messages (MESSAGE sip:100 at mydomain.com <sip:100 at mydomain.com>) also go through these dialplans? If no, how can I handle them?
>  
> This might be a stupid question but any help is still appreciated.
>  
>  
> Best Regards
> Dimitry Nagorny
> Trainee
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
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From abaci64 at gmail.com  Wed Jun  1 05:46:47 2016
From: abaci64 at gmail.com (Abaci B)
Date: Tue, 31 May 2016 21:46:47 -0400
Subject: [Freeswitch-users] Run a LUA script when voicemail received
In-Reply-To: <CAOkuKddVsmLA-+YjQo7myFF7HXta6K468t+7WooETYU6=_vJkw@mail.gmail.com>
References: <CAOkuKdfnwFsKN4H7ZxB+dvfGSGnAby59AM+kO4MCrLuJWa=Qag@mail.gmail.com>
	<CAEwTsAg0rPXMtvJtGns1kwcNWRT2rHJSnn+psXfnF9_BMdV3Ww@mail.gmail.com>
	<CAFmZFiNHye2Tujfwsw8mh9tc+bnZNxSbxTt670W0L3UtCKszGg@mail.gmail.com>
	<CAOkuKdczNFzL76oRF1f2SARJ7700owGmMnJOrtTZA6x=5AsLJQ@mail.gmail.com>
	<CAOkuKde_1pK6TzA4-9qKSVuh8A0s7i1qs2n4uUt-5-b8=HSgKQ@mail.gmail.com>
	<CAOkuKdcvRDVkmOhdMQ+7VB8edhdbzMX+25-wRAxgPSn-waX4iw@mail.gmail.com>
	<CAOkuKddVsmLA-+YjQo7myFF7HXta6K468t+7WooETYU6=_vJkw@mail.gmail.com>
Message-ID: <CAFmZFiN+q-MgFr6PfZEnDVe9hvqikASGwvCuS4A4orvX57RS7Q@mail.gmail.com>

This is the correct way to get user data, why do you consider this unclean?

On Tue, May 31, 2016 at 8:10 PM, Steven Schoch <
schoch+freeswitch.org at xwin32.com> wrote:

> I may have just answered my own question again. I found a way, but please
> let me know if you know of a cleaner way:
>
> user = event:getHeader("VM-User")
> domain = event:getHeader("VM-Domain")
> api = freeswitch.API();
> from = api:executeString("user_data " ..user .. "@" .. domain ..  " var
> outbound_caller_id_number")
> to = api:executeString("user_data " .. user .. "@" .. domain .. " param
> vm-notify-sms")
>
> This should get me what I need to send the text.
>
> --
> Steve
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From aubalde at presenceco.com  Wed Jun  1 17:50:27 2016
From: aubalde at presenceco.com (=?UTF-8?Q?Agust=C3=AD_Ubalde?=)
Date: Wed, 1 Jun 2016 15:50:27 +0200
Subject: [Freeswitch-users] Loop in the dialplan
Message-ID: <CAKThY06CLeWJ8q9vY4wS=SJtRxfpRnmr_gbCm8m8AkOXxk9oHw@mail.gmail.com>

Hi all,

How to simulate a loop in the dialplan? Like goto in Asterisk.


Thanks!

*PRESENCE TECHNOLOGY*
*Agust? Ubalde Bellot*
Chief Developer
C/ Comte Urgell 240 3A
Barcelona 08036
aubalde at presenceco.com

Ph: +34 93 10 10 300
Fx: +34 93 10 10 333

*www.presenceco.com* <http://www.presenceco.com/>

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-- 



*Presence Technology - DisclaimerThis message, its content and any file 
attached thereto is for the intended recipient only and is confidential and 
/or privileged. If you have received this e-mail in error or had access to 
it, you should note that the information in it is private and any use 
thereof is unauthorized. In such an event please notify us by e-mail or by 
telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by 
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From italo at freeswitch.org  Wed Jun  1 18:12:25 2016
From: italo at freeswitch.org (=?utf-8?Q?=C3=8Dtalo_Rossi?=)
Date: Wed, 1 Jun 2016 11:12:25 -0300
Subject: [Freeswitch-users] Loop in the dialplan
In-Reply-To: <CAKThY06CLeWJ8q9vY4wS=SJtRxfpRnmr_gbCm8m8AkOXxk9oHw@mail.gmail.com>
References: <CAKThY06CLeWJ8q9vY4wS=SJtRxfpRnmr_gbCm8m8AkOXxk9oHw@mail.gmail.com>
Message-ID: <C0287CC9-20CC-4D30-A4B8-4E70DFB9BFF6@freeswitch.org>

https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+transfer

> On 1 de jun de 2016, at 10:50, Agust? Ubalde <aubalde at presenceco.com> wrote:
> 
> Hi all,
> 
> How to simulate a loop in the dialplan? Like goto in Asterisk.
> 
> 
> Thanks!
> PRESENCE TECHNOLOGY
> Agust? Ubalde Bellot
> Chief Developer
> C/ Comte Urgell 240 3A
> Barcelona 08036
> aubalde at presenceco.com
> 
> Ph: +34 93 10 10 300 
> Fx: +34 93 10 10 333
> 
> 
> www.presenceco.com
> 
> Follow us on:
> 
> 
> 
> For additional information, please visit our website www.presenceco.com
> 
> 
> Presence Technology - Disclaimer
> This message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.
> For additional information, please visit our website www.presenceco.com
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

?talo Rossi
italo at freeswitch.org





From krice at freeswitch.org  Wed Jun  1 18:21:27 2016
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 1 Jun 2016 09:21:27 -0500
Subject: [Freeswitch-users] Loop in the dialplan
In-Reply-To: <CAKThY06CLeWJ8q9vY4wS=SJtRxfpRnmr_gbCm8m8AkOXxk9oHw@mail.gmail.com>
References: <CAKThY06CLeWJ8q9vY4wS=SJtRxfpRnmr_gbCm8m8AkOXxk9oHw@mail.gmail.com>
Message-ID: <C4104F60-B8D4-4D02-A311-1CD9CBB3FA40@freeswitch.org>

Freeswitch dialplan is not a scripting language. It is not interpreted one command at a time like asterisk dialplan is. If you need to script calls you should be looking at one of the scripting languages like lua, js (mod_v8) perl etc or use something external like httapi or xml curl 

Sent from my iPhone

> On Jun 1, 2016, at 8:50 AM, Agust? Ubalde <aubalde at presenceco.com> wrote:
> 
> Hi all,
> 
> How to simulate a loop in the dialplan? Like goto in Asterisk.
> 
> 
> Thanks!
> PRESENCE TECHNOLOGY
> Agust? Ubalde Bellot
> Chief Developer
> C/ Comte Urgell 240 3A
> Barcelona 08036
> aubalde at presenceco.com
> 
> Ph: +34 93 10 10 300 
> Fx: +34 93 10 10 333
> 
> 
> www.presenceco.com
> 
> Follow us on:
> 
> 
> 
> For additional information, please visit our website www.presenceco.com
> 
> 
> Presence Technology - Disclaimer
> This message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.
> For additional information, please visit our website www.presenceco.com
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From abaci64 at gmail.com  Wed Jun  1 20:04:22 2016
From: abaci64 at gmail.com (Abaci B)
Date: Wed, 1 Jun 2016 12:04:22 -0400
Subject: [Freeswitch-users] Bridge Call and Join Conference
In-Reply-To: <CAPtU-UqO_jVO1OFxthu7TkqWCitUpwcU2E3SU0p0QmpXJ09MMQ@mail.gmail.com>
References: <CAPtU-UqO_jVO1OFxthu7TkqWCitUpwcU2E3SU0p0QmpXJ09MMQ@mail.gmail.com>
Message-ID: <CAFmZFiMfBeH5XRa6KqFq27COr-h_YKWFKR4FQX6Vi-ej9WRPGQ@mail.gmail.com>

one way to do it would be to use api_on_answer
<https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-Theapi_onfamily>
to execute a uuid_transfer -both
<https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-uuid_transfer>
to transfer both legs to the conference (make sure to set
hangup_after_bridge to false).

On Tue, May 31, 2016 at 4:59 PM, Colin Morelli <colin.morelli at gmail.com>
wrote:

> So, I'm trying to setup certain calls such that they enter into a
> conference asap. I have the process working using bridging conferences
> right now, which is okay, but bridging conferences forces the dialing leg
> of the call to go to an answered state immediately, while the bridge leg is
> still ringing. So, while everything seems to work correctly, it messes with
> some of my reconciliation/reporting process with another database that are
> fed off of channel state events. Essentially it looks like every call was
> answered.
>
> So I was curious to know if there's a way to make a call that creates a
> bridged leg, and then as soon as the leg is answered immediately joins both
> legs into a conference. Is the only way to do it to listen for the
> channel_bridge events and then transfer the call myself?
>
> Best,
> Colin
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mirkobrankovic at gmail.com  Wed Jun  1 00:51:30 2016
From: mirkobrankovic at gmail.com (Mirko Brankovic)
Date: Tue, 31 May 2016 22:51:30 +0200
Subject: [Freeswitch-users] ReInvte in bypass/proxy mode | a > a+v
Message-ID: <CAND18T0Yh6w1paAfAyrv5SHhM+7WrWhMyh257HGRg5T8FQo=nw@mail.gmail.com>

Hi,

Can someone help me figure out the problem i am hitting.

Im trying to upgrade call from audio to audio + video (opus 48k and vp8-
but this one is only in passtrough mode).

So set up is browser (sip.js/node.js) >> kamailio wss >> freswitch (bypass
mode bridge without answer) >> kamailio >> browser (node.js)

Problem is that freeswitch receives reInvite on existing audio call with
new sdp containing also video codecs, but directly from client so no proxy
(kamailio) in path, and exiting invite towards kamailio for outgoing leg is
missing video in a=group BUNDLE, so only audio and a=sendrcv is also being
stripped (that is the difference to 1 on 1 video call).

I understood that bypass mode shouldn't change/touch sdp.
Also passtrough codec shouldn't be the problem i guess since both clients
can decode it.

Any help is appreciated.
Thanks,
Mirko
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From william at magicwilly.info  Wed Jun  1 08:11:38 2016
From: william at magicwilly.info (William Scott)
Date: Wed, 1 Jun 2016 14:11:38 +1000
Subject: [Freeswitch-users] SwitchPi
Message-ID: <CABV9BUu8ygpbbGLzxmzG70NsiS7dj5D4WRkNaQH6Hzex_Q38yg@mail.gmail.com>

Is the SwitchPi project and blog still happening?
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From deepikay at iiitd.ac.in  Wed Jun  1 20:05:55 2016
From: deepikay at iiitd.ac.in (Deepika Yadav)
Date: Wed, 1 Jun 2016 21:35:55 +0530
Subject: [Freeswitch-users] Freeswitch Inbound Calls No sound
Message-ID: <CAM99_Y=Fj=ap7vybE5Yquws+06n_P=i-azesyfj4V3X0kag56w@mail.gmail.com>

Hi,

I need to make calls from a mobile number that can be answered by
Freeswitch extension playing an IVR.

To achieve this, I have configured a VOIP-GSM gateway having inserted a
mobile sim that upon receiving the incoming calls forwards it to a SIP URI.

Gateway Settings:

PBX IP : ec2 public ip
Protocol : SIP
Port : 5080

Target: 9811806537


Freesswitch resides at Amazon ec2 cloud,

Dialplan for public context is :

<extension name="mydid">
    <condition field="destination_number" expression="^(9811806537)$">
     <action application="transfer" data="4444 XML default"/>
    </condition>
  </extension>

Default Context :

<extension name="ivr_demo">
      <condition field="destination_number" expression="^4444$">
        <action application="answer"/>
        <action application="start_dtmf" />
        <action application="sleep" data="2000"/>
        <action application="ivr" data="rhd_ivr"/>
      </condition>
</extension>


where Server IP - XX.XX.XX.XX
Gateway IP- YY.YY.YY.YY


Freeswitch logs can be found here :

http://pastebin.com/7BbM5dWD <http://pastebin.com/i4gs9uVJ>

Files
sip_profiles/external.xml
sip_profiles/internal.xml

<param name="ext-rtp-ip" value="XX.XX.XX.XX"/>
<param name="ext-sip-ip" value="XX.XX.XX.XX"/>


Any pointer to move forward would be highly appreciated

Reagrds,
Deepika
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From stevem at singlecomm.com  Wed Jun  1 20:09:55 2016
From: stevem at singlecomm.com (stevem)
Date: Wed, 1 Jun 2016 09:09:55 -0700 (MST)
Subject: [Freeswitch-users] Webrtc Verto jsapi
Message-ID: <1464797395361-7596220.post@n2.nabble.com>

I was not able to find any documentation on this. Basically all I want to do
from my frontend. Is post an event to the backend. Basically run the
command: uuid_broadcast 69d58330-7f7f-4e52-bb2c-9e25d9f5238b
event::Event-Name=HOLD.

I'm using FreeSWITCH Version 1.4.21~64bit

I got the verto/webrtc working using the demo located here:
https://webrtc.freeswitch.org/verto/index.html#page-main

I'm able to connect to the dialplan and drop the caller (PSTN/ITSPs) and
verto client into a conference and the audio is flowing. I just want to send
a message to the backend to tell it to do things. In my example above I want
to set the caller on hold.

My verto connection in javascript looks like this:

    $.verto.init({}, bootstrap);

    function bootstrap(status) {
      that.vertoHandle = new jQuery.verto({
        login: userId + '@' + host,
        passwd: password,
        tag: "vertoAudio",
        socketUrl: 'wss://' + host + ':8082',
        audioParams: {
          googAutoGainControl: false,
          googNoiseSuppression: false,
          googHighpassFilter: false
        },
        iceServers: true
      }, vertoCallbacks);
    }

    var vertoCallbacks = {
      onWSLogin: onWSLogin,
      onWSClose: onWSClose,
      onDialogState: onDialogState
    };


On successful login i make a call:

     that.call= that.vertoHandle.newCall({
          destination_number: that.liveCall.ani,
          caller_id_name: user.firstName + ' ' + user.lastName,
          caller_id_number: userId,
          useVideo: false,
          useStereo: false,
          userVariables: {}
        });


Thanks alot. This is my first post and I've only been doing Freeswitch for a
few months. I have alot of experience in Asterisk though. Let me know if
more data is needed.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Webrtc-Verto-jsapi-tp7596220.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From colin.morelli at gmail.com  Wed Jun  1 20:13:37 2016
From: colin.morelli at gmail.com (Colin Morelli)
Date: Wed, 01 Jun 2016 16:13:37 +0000
Subject: [Freeswitch-users] Bridge Call and Join Conference
In-Reply-To: <CAFmZFiMfBeH5XRa6KqFq27COr-h_YKWFKR4FQX6Vi-ej9WRPGQ@mail.gmail.com>
References: <CAPtU-UqO_jVO1OFxthu7TkqWCitUpwcU2E3SU0p0QmpXJ09MMQ@mail.gmail.com>
	<CAFmZFiMfBeH5XRa6KqFq27COr-h_YKWFKR4FQX6Vi-ej9WRPGQ@mail.gmail.com>
Message-ID: <CAPtU-UpJMabAq_ngz6wpFScigO834o-ACw5QmgK1mLNrMcJJ_w@mail.gmail.com>

Awesome - this is exactly what I was looking for. Don't know how I missed
it.

Thank you!

Best,
Colin

On Wed, Jun 1, 2016 at 12:08 PM Abaci B <abaci64 at gmail.com> wrote:

> one way to do it would be to use api_on_answer
> <https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-Theapi_onfamily>
> to execute a uuid_transfer -both
> <https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-uuid_transfer>
> to transfer both legs to the conference (make sure to set
> hangup_after_bridge to false).
>
> On Tue, May 31, 2016 at 4:59 PM, Colin Morelli <colin.morelli at gmail.com>
> wrote:
>
>> So, I'm trying to setup certain calls such that they enter into a
>> conference asap. I have the process working using bridging conferences
>> right now, which is okay, but bridging conferences forces the dialing leg
>> of the call to go to an answered state immediately, while the bridge leg is
>> still ringing. So, while everything seems to work correctly, it messes with
>> some of my reconciliation/reporting process with another database that are
>> fed off of channel state events. Essentially it looks like every call was
>> answered.
>>
>> So I was curious to know if there's a way to make a call that creates a
>> bridged leg, and then as soon as the leg is answered immediately joins both
>> legs into a conference. Is the only way to do it to listen for the
>> channel_bridge events and then transfer the call myself?
>>
>> Best,
>> Colin
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From krice at freeswitch.org  Wed Jun  1 20:14:25 2016
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 1 Jun 2016 11:14:25 -0500
Subject: [Freeswitch-users] SwitchPi
In-Reply-To: <CABV9BUu8ygpbbGLzxmzG70NsiS7dj5D4WRkNaQH6Hzex_Q38yg@mail.gmail.com>
References: <CABV9BUu8ygpbbGLzxmzG70NsiS7dj5D4WRkNaQH6Hzex_Q38yg@mail.gmail.com>
Message-ID: <00bb01d1bc20$ab399cb0$01acd610$@freeswitch.org>

Its still out there? the site is dead? but FS still runs on RPi, the drivers are still on github, and I?ll have a pile of the little custom hardware boards for the RPi @ ClueCon

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William Scott
Sent: Tuesday, May 31, 2016 11:12 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] SwitchPi

 

Is the SwitchPi project and blog still happening?




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From yu at yu-boot.ru  Wed Jun  1 21:26:45 2016
From: yu at yu-boot.ru (Yu Boot)
Date: Wed, 01 Jun 2016 20:26:45 +0300
Subject: [Freeswitch-users] SIP gateways, context
Message-ID: <e5w3xs40kl8n2vqlk9f2xxjr.1464801824356@email.android.com>

I use single profile both for SIP gateways and for password-protected clients. Why FS ignores context parameter in SIP gateway settings? I can route inbound calls from these SIP trunks only in "default" context by inbound IP address. Password-protected clients (from directory) don't suffer from this, all inbound calls are routed via respective contexts.

Sent from my ASUS
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From olegstolyar at gmail.com  Wed Jun  1 23:40:22 2016
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Wed, 1 Jun 2016 12:40:22 -0700
Subject: [Freeswitch-users] Is there a way for FS not to send back any
	response to INVITE
Message-ID: <CAC8JMOKzRZYC1z58qEcqBTHXDnneMbbD7M5Hmv_JUvD9MZU5Sw@mail.gmail.com>

Hi,

In order to protect against scanning attacks I'd like for FS to not respond
to INVITES unless they match certain conditions.

I understand that currently FS always responds with 100 Trying right away
before processing the call and then, if the call does not match anything in
the dialplan, responds with a 302 Moved Temporarily.

The 302 can be replaced with another response code (for example 403
Forbidden which is what I am doing now) using the *respond* dialplan app.
However, that might encourage the scanner to keep trying.

So I guess there are two questions:

1. Is there a way not to send back 100 Trying at all?

2. Is there a way to not send any final response?
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From italo at freeswitch.org  Wed Jun  1 23:46:13 2016
From: italo at freeswitch.org (=?utf-8?Q?=C3=8Dtalo_Rossi?=)
Date: Wed, 1 Jun 2016 16:46:13 -0300
Subject: [Freeswitch-users] Webrtc Verto jsapi
In-Reply-To: <1464797395361-7596220.post@n2.nabble.com>
References: <1464797395361-7596220.post@n2.nabble.com>
Message-ID: <E05085C5-D0A7-4DBF-A5E7-CC414DF3070E@freeswitch.org>

http://evoluxbr.github.io/verto-docs/tut/holding-a-call.html


> On 1 de jun de 2016, at 13:09, stevem <stevem at singlecomm.com> wrote:
> 
> I was not able to find any documentation on this. Basically all I want to do
> from my frontend. Is post an event to the backend. Basically run the
> command: uuid_broadcast 69d58330-7f7f-4e52-bb2c-9e25d9f5238b
> event::Event-Name=HOLD.
> 
> I'm using FreeSWITCH Version 1.4.21~64bit
> 
> I got the verto/webrtc working using the demo located here:
> https://webrtc.freeswitch.org/verto/index.html#page-main
> 
> I'm able to connect to the dialplan and drop the caller (PSTN/ITSPs) and
> verto client into a conference and the audio is flowing. I just want to send
> a message to the backend to tell it to do things. In my example above I want
> to set the caller on hold.
> 
> My verto connection in javascript looks like this:
> 
>    $.verto.init({}, bootstrap);
> 
>    function bootstrap(status) {
>      that.vertoHandle = new jQuery.verto({
>        login: userId + '@' + host,
>        passwd: password,
>        tag: "vertoAudio",
>        socketUrl: 'wss://' + host + ':8082',
>        audioParams: {
>          googAutoGainControl: false,
>          googNoiseSuppression: false,
>          googHighpassFilter: false
>        },
>        iceServers: true
>      }, vertoCallbacks);
>    }
> 
>    var vertoCallbacks = {
>      onWSLogin: onWSLogin,
>      onWSClose: onWSClose,
>      onDialogState: onDialogState
>    };
> 
> 
> On successful login i make a call:
> 
>     that.call= that.vertoHandle.newCall({
>          destination_number: that.liveCall.ani,
>          caller_id_name: user.firstName + ' ' + user.lastName,
>          caller_id_number: userId,
>          useVideo: false,
>          useStereo: false,
>          userVariables: {}
>        });
> 
> 
> Thanks alot. This is my first post and I've only been doing Freeswitch for a
> few months. I have alot of experience in Asterisk though. Let me know if
> more data is needed.
> 
> 
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Webrtc-Verto-jsapi-tp7596220.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

?talo Rossi
italo at freeswitch.org





From italo at freeswitch.org  Wed Jun  1 23:51:46 2016
From: italo at freeswitch.org (=?utf-8?Q?=C3=8Dtalo_Rossi?=)
Date: Wed, 1 Jun 2016 16:51:46 -0300
Subject: [Freeswitch-users] Is there a way for FS not to send back any
	response to INVITE
In-Reply-To: <CAC8JMOKzRZYC1z58qEcqBTHXDnneMbbD7M5Hmv_JUvD9MZU5Sw@mail.gmail.com>
References: <CAC8JMOKzRZYC1z58qEcqBTHXDnneMbbD7M5Hmv_JUvD9MZU5Sw@mail.gmail.com>
Message-ID: <9AD84DF4-B4C0-4732-9D58-118F84439013@freeswitch.org>

I think you can write a dialplan module for the first question. Mike or Anthony can confirm this.


> On 1 de jun de 2016, at 16:40, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> 
> Hi,
> 
> In order to protect against scanning attacks I'd like for FS to not respond to INVITES unless they match certain conditions.  
> 
> I understand that currently FS always responds with 100 Trying right away before processing the call and then, if the call does not match anything in the dialplan, responds with a 302 Moved Temporarily.
> 
> The 302 can be replaced with another response code (for example 403 Forbidden which is what I am doing now) using the respond dialplan app.   However, that might encourage the scanner to keep trying.
> 
> So I guess there are two questions:
> 
> 1. Is there a way not to send back 100 Trying at all?
> 
> 2. Is there a way to not send any final response?
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

?talo Rossi
italo at freeswitch.org





From s.safarov at gmail.com  Wed Jun  1 23:59:04 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Wed, 01 Jun 2016 19:59:04 +0000
Subject: [Freeswitch-users] ReInvte in bypass/proxy mode | a > a+v
In-Reply-To: <CAND18T0Yh6w1paAfAyrv5SHhM+7WrWhMyh257HGRg5T8FQo=nw@mail.gmail.com>
References: <CAND18T0Yh6w1paAfAyrv5SHhM+7WrWhMyh257HGRg5T8FQo=nw@mail.gmail.com>
Message-ID: <CAHtxdDckjrTpjQmyhcJCZKfOnSD6Ar=4HzJpZTMx7KHTRuZYyA@mail.gmail.com>

Look at FS-9203 <https://freeswitch.org/jira/browse/FS-9203> and FS-9206
<https://freeswitch.org/jira/browse/FS-9206>


??, 1 ???. 2016 ?. ? 19:12, Mirko Brankovic <mirkobrankovic at gmail.com>:

> Hi,
>
> Can someone help me figure out the problem i am hitting.
>
> Im trying to upgrade call from audio to audio + video (opus 48k and vp8-
> but this one is only in passtrough mode).
>
> So set up is browser (sip.js/node.js) >> kamailio wss >> freswitch (bypass
> mode bridge without answer) >> kamailio >> browser (node.js)
>
> Problem is that freeswitch receives reInvite on existing audio call with
> new sdp containing also video codecs, but directly from client so no proxy
> (kamailio) in path, and exiting invite towards kamailio for outgoing leg is
> missing video in a=group BUNDLE, so only audio and a=sendrcv is also being
> stripped (that is the difference to 1 on 1 video call).
>
> I understood that bypass mode shouldn't change/touch sdp.
> Also passtrough codec shouldn't be the problem i guess since both clients
> can decode it.
>
> Any help is appreciated.
> Thanks,
> Mirko
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From stevem at singlecomm.com  Wed Jun  1 23:59:18 2016
From: stevem at singlecomm.com (Steve Mathers)
Date: Wed, 1 Jun 2016 13:59:18 -0600
Subject: [Freeswitch-users] Webrtc Verto jsapi
In-Reply-To: <E05085C5-D0A7-4DBF-A5E7-CC414DF3070E@freeswitch.org>
References: <1464797395361-7596220.post@n2.nabble.com>
	<E05085C5-D0A7-4DBF-A5E7-CC414DF3070E@freeswitch.org>
Message-ID: <CABhxEyKOUjtLUUHy0eO-juQQv_v1uuwpJXv2a_COSTpHapBXkA@mail.gmail.com>

Thanks, I did see this. But not what i'm looking for. I still want to be
able to send commands to the backend. In my case I want to broadcast a
specific event to a specific channel.

On Wed, Jun 1, 2016 at 1:46 PM, ?talo Rossi <italo at freeswitch.org> wrote:

> http://evoluxbr.github.io/verto-docs/tut/holding-a-call.html
>
>
> > On 1 de jun de 2016, at 13:09, stevem <stevem at singlecomm.com> wrote:
> >
> > I was not able to find any documentation on this. Basically all I want
> to do
> > from my frontend. Is post an event to the backend. Basically run the
> > command: uuid_broadcast 69d58330-7f7f-4e52-bb2c-9e25d9f5238b
> > event::Event-Name=HOLD.
> >
> > I'm using FreeSWITCH Version 1.4.21~64bit
> >
> > I got the verto/webrtc working using the demo located here:
> > https://webrtc.freeswitch.org/verto/index.html#page-main
> >
> > I'm able to connect to the dialplan and drop the caller (PSTN/ITSPs) and
> > verto client into a conference and the audio is flowing. I just want to
> send
> > a message to the backend to tell it to do things. In my example above I
> want
> > to set the caller on hold.
> >
> > My verto connection in javascript looks like this:
> >
> >    $.verto.init({}, bootstrap);
> >
> >    function bootstrap(status) {
> >      that.vertoHandle = new jQuery.verto({
> >        login: userId + '@' + host,
> >        passwd: password,
> >        tag: "vertoAudio",
> >        socketUrl: 'wss://' + host + ':8082',
> >        audioParams: {
> >          googAutoGainControl: false,
> >          googNoiseSuppression: false,
> >          googHighpassFilter: false
> >        },
> >        iceServers: true
> >      }, vertoCallbacks);
> >    }
> >
> >    var vertoCallbacks = {
> >      onWSLogin: onWSLogin,
> >      onWSClose: onWSClose,
> >      onDialogState: onDialogState
> >    };
> >
> >
> > On successful login i make a call:
> >
> >     that.call= that.vertoHandle.newCall({
> >          destination_number: that.liveCall.ani,
> >          caller_id_name: user.firstName + ' ' + user.lastName,
> >          caller_id_number: userId,
> >          useVideo: false,
> >          useStereo: false,
> >          userVariables: {}
> >        });
> >
> >
> > Thanks alot. This is my first post and I've only been doing Freeswitch
> for a
> > few months. I have alot of experience in Asterisk though. Let me know if
> > more data is needed.
> >
> >
> >
> > --
> > View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Webrtc-Verto-jsapi-tp7596220.html
> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> ?talo Rossi
> italo at freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mike at jerris.com  Thu Jun  2 00:00:14 2016
From: mike at jerris.com (Michael Jerris)
Date: Wed, 1 Jun 2016 16:00:14 -0400
Subject: [Freeswitch-users] Is there a way for FS not to send back any
	response to INVITE
In-Reply-To: <CAC8JMOKzRZYC1z58qEcqBTHXDnneMbbD7M5Hmv_JUvD9MZU5Sw@mail.gmail.com>
References: <CAC8JMOKzRZYC1z58qEcqBTHXDnneMbbD7M5Hmv_JUvD9MZU5Sw@mail.gmail.com>
Message-ID: <CDCB6849-263E-4595-9EAD-05911832ECF3@jerris.com>

The only way with our current sip module to accomplish either of these would be to put a sip proxy out front to handle that behavior, or to somehow use iptables to block the traffic

> On Jun 1, 2016, at 3:40 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> 
> Hi,
> 
> In order to protect against scanning attacks I'd like for FS to not respond to INVITES unless they match certain conditions.  
> 
> I understand that currently FS always responds with 100 Trying right away before processing the call and then, if the call does not match anything in the dialplan, responds with a 302 Moved Temporarily.
> 
> The 302 can be replaced with another response code (for example 403 Forbidden which is what I am doing now) using the respond dialplan app.   However, that might encourage the scanner to keep trying.
> 
> So I guess there are two questions:
> 
> 1. Is there a way not to send back 100 Trying at all?
> 
> 2. Is there a way to not send any final response?

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From olegstolyar at gmail.com  Thu Jun  2 00:05:26 2016
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Wed, 1 Jun 2016 13:05:26 -0700
Subject: [Freeswitch-users] Is there a way for FS not to send back any
 response to INVITE
In-Reply-To: <CDCB6849-263E-4595-9EAD-05911832ECF3@jerris.com>
References: <CAC8JMOKzRZYC1z58qEcqBTHXDnneMbbD7M5Hmv_JUvD9MZU5Sw@mail.gmail.com>
	<CDCB6849-263E-4595-9EAD-05911832ECF3@jerris.com>
Message-ID: <CAC8JMO+4sQduNMf3-eAve+UFHHBoiHnEBd76S6NO1o3=KfBuSw@mail.gmail.com>

Thanks guys!  IP tables is how we block most traffic but we can only block
traffic by port.  In this case it's about invalid INVITES coming in on a
valid port.

Do you think this functionality would be useful?
Is it worth opening a feature request and perhaps putting a bounty on it?
Any idea of the effort?

On Wed, Jun 1, 2016 at 1:00 PM, Michael Jerris <mike at jerris.com> wrote:

> The only way with our current sip module to accomplish either of these
> would be to put a sip proxy out front to handle that behavior, or to
> somehow use iptables to block the traffic
>
> On Jun 1, 2016, at 3:40 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>
> Hi,
>
> In order to protect against scanning attacks I'd like for FS to not
> respond to INVITES unless they match certain conditions.
>
> I understand that currently FS always responds with 100 Trying right away
> before processing the call and then, if the call does not match anything in
> the dialplan, responds with a 302 Moved Temporarily.
>
> The 302 can be replaced with another response code (for example 403
> Forbidden which is what I am doing now) using the *respond* dialplan app.
>   However, that might encourage the scanner to keep trying.
>
> So I guess there are two questions:
>
> 1. Is there a way not to send back 100 Trying at all?
>
> 2. Is there a way to not send any final response?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Thu Jun  2 00:11:39 2016
From: mike at jerris.com (Michael Jerris)
Date: Wed, 1 Jun 2016 16:11:39 -0400
Subject: [Freeswitch-users] Is there a way for FS not to send back any
	response to INVITE
In-Reply-To: <CAC8JMO+4sQduNMf3-eAve+UFHHBoiHnEBd76S6NO1o3=KfBuSw@mail.gmail.com>
References: <CAC8JMOKzRZYC1z58qEcqBTHXDnneMbbD7M5Hmv_JUvD9MZU5Sw@mail.gmail.com>
	<CDCB6849-263E-4595-9EAD-05911832ECF3@jerris.com>
	<CAC8JMO+4sQduNMf3-eAve+UFHHBoiHnEBd76S6NO1o3=KfBuSw@mail.gmail.com>
Message-ID: <FFA66A9C-9986-4863-BD1F-39494AC86FD8@jerris.com>

If you want to do this, put a proxy out in front... Doing it in freeswitch would require a complete rewrite of the sip module.

> On Jun 1, 2016, at 4:05 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> 
> Thanks guys!  IP tables is how we block most traffic but we can only block traffic by port.  In this case it's about invalid INVITES coming in on a valid port.
> 
> Do you think this functionality would be useful?  
> Is it worth opening a feature request and perhaps putting a bounty on it?  
> Any idea of the effort?
> 
> On Wed, Jun 1, 2016 at 1:00 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> The only way with our current sip module to accomplish either of these would be to put a sip proxy out front to handle that behavior, or to somehow use iptables to block the traffic
> 
>> On Jun 1, 2016, at 3:40 PM, Oleg Stolyar <olegstolyar at gmail.com <mailto:olegstolyar at gmail.com>> wrote:
>> 
>> Hi,
>> 
>> In order to protect against scanning attacks I'd like for FS to not respond to INVITES unless they match certain conditions.  
>> 
>> I understand that currently FS always responds with 100 Trying right away before processing the call and then, if the call does not match anything in the dialplan, responds with a 302 Moved Temporarily.
>> 
>> The 302 can be replaced with another response code (for example 403 Forbidden which is what I am doing now) using the respond dialplan app.   However, that might encourage the scanner to keep trying.
>> 
>> So I guess there are two questions:
>> 
>> 1. Is there a way not to send back 100 Trying at all?
>> 
>> 2. Is there a way to not send any final response?
> 

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From jurijs.ivolga at gmail.com  Thu Jun  2 00:26:48 2016
From: jurijs.ivolga at gmail.com (Jurijs Ivolga)
Date: Wed, 1 Jun 2016 23:26:48 +0300
Subject: [Freeswitch-users] Is there a way for FS not to send back any
 response to INVITE
In-Reply-To: <CAC8JMO+4sQduNMf3-eAve+UFHHBoiHnEBd76S6NO1o3=KfBuSw@mail.gmail.com>
References: <CAC8JMOKzRZYC1z58qEcqBTHXDnneMbbD7M5Hmv_JUvD9MZU5Sw@mail.gmail.com>
	<CDCB6849-263E-4595-9EAD-05911832ECF3@jerris.com>
	<CAC8JMO+4sQduNMf3-eAve+UFHHBoiHnEBd76S6NO1o3=KfBuSw@mail.gmail.com>
Message-ID: <CAOrA2UaEmoT+2u-6u0uKG9Y5x+ajijEPQWgaa2V9LSRe0V7x_Q@mail.gmail.com>

Hi Oleg,

With iptables you can block based on what is inside SIP packet(off cause if
you are not using TLS), take a look on link below:

http://www.bertera.it/index.php/2014/01/22/sip-facket-filtering-with-iptables/

It is not best way to achieve what you need, cause as far as I know it is
resource consuming operations. Best way will be to use Kamailio as SIP
proxy in front.

With kind regards,

Jurijs

On Wed, Jun 1, 2016 at 11:05 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:

> Thanks guys!  IP tables is how we block most traffic but we can only block
> traffic by port.  In this case it's about invalid INVITES coming in on a
> valid port.
>
> Do you think this functionality would be useful?
> Is it worth opening a feature request and perhaps putting a bounty on it?
> Any idea of the effort?
>
> On Wed, Jun 1, 2016 at 1:00 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> The only way with our current sip module to accomplish either of these
>> would be to put a sip proxy out front to handle that behavior, or to
>> somehow use iptables to block the traffic
>>
>> On Jun 1, 2016, at 3:40 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>>
>> Hi,
>>
>> In order to protect against scanning attacks I'd like for FS to not
>> respond to INVITES unless they match certain conditions.
>>
>> I understand that currently FS always responds with 100 Trying right away
>> before processing the call and then, if the call does not match anything in
>> the dialplan, responds with a 302 Moved Temporarily.
>>
>> The 302 can be replaced with another response code (for example 403
>> Forbidden which is what I am doing now) using the *respond* dialplan
>> app.   However, that might encourage the scanner to keep trying.
>>
>> So I guess there are two questions:
>>
>> 1. Is there a way not to send back 100 Trying at all?
>>
>> 2. Is there a way to not send any final response?
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From olegstolyar at gmail.com  Thu Jun  2 00:27:27 2016
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Wed, 1 Jun 2016 13:27:27 -0700
Subject: [Freeswitch-users] Is there a way for FS not to send back any
 response to INVITE
In-Reply-To: <FFA66A9C-9986-4863-BD1F-39494AC86FD8@jerris.com>
References: <CAC8JMOKzRZYC1z58qEcqBTHXDnneMbbD7M5Hmv_JUvD9MZU5Sw@mail.gmail.com>
	<CDCB6849-263E-4595-9EAD-05911832ECF3@jerris.com>
	<CAC8JMO+4sQduNMf3-eAve+UFHHBoiHnEBd76S6NO1o3=KfBuSw@mail.gmail.com>
	<FFA66A9C-9986-4863-BD1F-39494AC86FD8@jerris.com>
Message-ID: <CAC8JMOLukCN1t=08LQYV8VKO_F4VNUmNgxXWwGH-yb5idAg7iw@mail.gmail.com>

Got it, thanks Michael!

On Wed, Jun 1, 2016 at 1:11 PM, Michael Jerris <mike at jerris.com> wrote:

> If you want to do this, put a proxy out in front... Doing it in freeswitch
> would require a complete rewrite of the sip module.
>
> On Jun 1, 2016, at 4:05 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>
> Thanks guys!  IP tables is how we block most traffic but we can only block
> traffic by port.  In this case it's about invalid INVITES coming in on a
> valid port.
>
> Do you think this functionality would be useful?
> Is it worth opening a feature request and perhaps putting a bounty on it?
> Any idea of the effort?
>
> On Wed, Jun 1, 2016 at 1:00 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> The only way with our current sip module to accomplish either of these
>> would be to put a sip proxy out front to handle that behavior, or to
>> somehow use iptables to block the traffic
>>
>> On Jun 1, 2016, at 3:40 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>>
>> Hi,
>>
>> In order to protect against scanning attacks I'd like for FS to not
>> respond to INVITES unless they match certain conditions.
>>
>> I understand that currently FS always responds with 100 Trying right away
>> before processing the call and then, if the call does not match anything in
>> the dialplan, responds with a 302 Moved Temporarily.
>>
>> The 302 can be replaced with another response code (for example 403
>> Forbidden which is what I am doing now) using the *respond* dialplan
>> app.   However, that might encourage the scanner to keep trying.
>>
>> So I guess there are two questions:
>>
>> 1. Is there a way not to send back 100 Trying at all?
>>
>> 2. Is there a way to not send any final response?
>>
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From olegstolyar at gmail.com  Thu Jun  2 00:38:47 2016
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Wed, 1 Jun 2016 13:38:47 -0700
Subject: [Freeswitch-users] Is there a way for FS not to send back any
 response to INVITE
In-Reply-To: <CAOrA2UaEmoT+2u-6u0uKG9Y5x+ajijEPQWgaa2V9LSRe0V7x_Q@mail.gmail.com>
References: <CAC8JMOKzRZYC1z58qEcqBTHXDnneMbbD7M5Hmv_JUvD9MZU5Sw@mail.gmail.com>
	<CDCB6849-263E-4595-9EAD-05911832ECF3@jerris.com>
	<CAC8JMO+4sQduNMf3-eAve+UFHHBoiHnEBd76S6NO1o3=KfBuSw@mail.gmail.com>
	<CAOrA2UaEmoT+2u-6u0uKG9Y5x+ajijEPQWgaa2V9LSRe0V7x_Q@mail.gmail.com>
Message-ID: <CAC8JMOJnvohg3T6-pb_wBqDAg6L5iUVhEjFPnoKHiNMSQBei9g@mail.gmail.com>

Thanks Jurijs!

Unfortunately we do need to use TLS.

On Wed, Jun 1, 2016 at 1:26 PM, Jurijs Ivolga <jurijs.ivolga at gmail.com>
wrote:

> Hi Oleg,
>
> With iptables you can block based on what is inside SIP packet(off cause
> if you are not using TLS), take a look on link below:
>
>
> http://www.bertera.it/index.php/2014/01/22/sip-facket-filtering-with-iptables/
>
> It is not best way to achieve what you need, cause as far as I know it is
> resource consuming operations. Best way will be to use Kamailio as SIP
> proxy in front.
>
> With kind regards,
>
> Jurijs
>
> On Wed, Jun 1, 2016 at 11:05 PM, Oleg Stolyar <olegstolyar at gmail.com>
> wrote:
>
>> Thanks guys!  IP tables is how we block most traffic but we can only
>> block traffic by port.  In this case it's about invalid INVITES coming in
>> on a valid port.
>>
>> Do you think this functionality would be useful?
>> Is it worth opening a feature request and perhaps putting a bounty on it?
>>
>> Any idea of the effort?
>>
>> On Wed, Jun 1, 2016 at 1:00 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> The only way with our current sip module to accomplish either of these
>>> would be to put a sip proxy out front to handle that behavior, or to
>>> somehow use iptables to block the traffic
>>>
>>> On Jun 1, 2016, at 3:40 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>>>
>>> Hi,
>>>
>>> In order to protect against scanning attacks I'd like for FS to not
>>> respond to INVITES unless they match certain conditions.
>>>
>>> I understand that currently FS always responds with 100 Trying right
>>> away before processing the call and then, if the call does not match
>>> anything in the dialplan, responds with a 302 Moved Temporarily.
>>>
>>> The 302 can be replaced with another response code (for example 403
>>> Forbidden which is what I am doing now) using the *respond* dialplan
>>> app.   However, that might encourage the scanner to keep trying.
>>>
>>> So I guess there are two questions:
>>>
>>> 1. Is there a way not to send back 100 Trying at all?
>>>
>>> 2. Is there a way to not send any final response?
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From krice at freeswitch.org  Thu Jun  2 00:52:06 2016
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 1 Jun 2016 15:52:06 -0500
Subject: [Freeswitch-users] Is there a way for FS not to send back any
	response to INVITE
In-Reply-To: <CAC8JMOJnvohg3T6-pb_wBqDAg6L5iUVhEjFPnoKHiNMSQBei9g@mail.gmail.com>
References: <CAC8JMOKzRZYC1z58qEcqBTHXDnneMbbD7M5Hmv_JUvD9MZU5Sw@mail.gmail.com>	<CDCB6849-263E-4595-9EAD-05911832ECF3@jerris.com>	<CAC8JMO+4sQduNMf3-eAve+UFHHBoiHnEBd76S6NO1o3=KfBuSw@mail.gmail.com>	<CAOrA2UaEmoT+2u-6u0uKG9Y5x+ajijEPQWgaa2V9LSRe0V7x_Q@mail.gmail.com>
	<CAC8JMOJnvohg3T6-pb_wBqDAg6L5iUVhEjFPnoKHiNMSQBei9g@mail.gmail.com>
Message-ID: <00cc01d1bc47$7600f850$6202e8f0$@freeswitch.org>

Keep in mind that with SIP if the server is listening you are supposed to respond. Not responding is a violation of the RFC? now that being side? using DPI via IPTables is a perfect way to dissuade the scanners? and btw, if you are using TLS, they don?t even need to see SIP to know you have something listening on TCP on the SIP port now, you?re syn-ack in reply to their syn already told them that?. 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Oleg Stolyar
Sent: Wednesday, June 1, 2016 3:39 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Is there a way for FS not to send back any response to INVITE

 

Thanks Jurijs!

 

Unfortunately we do need to use TLS.

 

On Wed, Jun 1, 2016 at 1:26 PM, Jurijs Ivolga <jurijs.ivolga at gmail.com <mailto:jurijs.ivolga at gmail.com> > wrote:

Hi Oleg,

With iptables you can block based on what is inside SIP packet(off cause if you are not using TLS), take a look on link below:

http://www.bertera.it/index.php/2014/01/22/sip-facket-filtering-with-iptables/

It is not best way to achieve what you need, cause as far as I know it is resource consuming operations. Best way will be to use Kamailio as SIP proxy in front.

With kind regards,




Jurijs

 

On Wed, Jun 1, 2016 at 11:05 PM, Oleg Stolyar <olegstolyar at gmail.com <mailto:olegstolyar at gmail.com> > wrote:

Thanks guys!  IP tables is how we block most traffic but we can only block traffic by port.  In this case it's about invalid INVITES coming in on a valid port.

 

Do you think this functionality would be useful?  

Is it worth opening a feature request and perhaps putting a bounty on it?  

Any idea of the effort?

 

On Wed, Jun 1, 2016 at 1:00 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com> > wrote:

The only way with our current sip module to accomplish either of these would be to put a sip proxy out front to handle that behavior, or to somehow use iptables to block the traffic

 

On Jun 1, 2016, at 3:40 PM, Oleg Stolyar <olegstolyar at gmail.com <mailto:olegstolyar at gmail.com> > wrote:

 

Hi,

 

In order to protect against scanning attacks I'd like for FS to not respond to INVITES unless they match certain conditions.  

 

I understand that currently FS always responds with 100 Trying right away before processing the call and then, if the call does not match anything in the dialplan, responds with a 302 Moved Temporarily.

 

The 302 can be replaced with another response code (for example 403 Forbidden which is what I am doing now) using the respond dialplan app.   However, that might encourage the scanner to keep trying.

 

So I guess there are two questions:

 

1. Is there a way not to send back 100 Trying at all?

 

2. Is there a way to not send any final response?

 

 

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

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

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

 


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

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

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

 


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

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

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

 

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From olegstolyar at gmail.com  Thu Jun  2 01:08:22 2016
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Wed, 1 Jun 2016 14:08:22 -0700
Subject: [Freeswitch-users] Is there a way for FS not to send back any
 response to INVITE
In-Reply-To: <00cc01d1bc47$7600f850$6202e8f0$@freeswitch.org>
References: <CAC8JMOKzRZYC1z58qEcqBTHXDnneMbbD7M5Hmv_JUvD9MZU5Sw@mail.gmail.com>
	<CDCB6849-263E-4595-9EAD-05911832ECF3@jerris.com>
	<CAC8JMO+4sQduNMf3-eAve+UFHHBoiHnEBd76S6NO1o3=KfBuSw@mail.gmail.com>
	<CAOrA2UaEmoT+2u-6u0uKG9Y5x+ajijEPQWgaa2V9LSRe0V7x_Q@mail.gmail.com>
	<CAC8JMOJnvohg3T6-pb_wBqDAg6L5iUVhEjFPnoKHiNMSQBei9g@mail.gmail.com>
	<00cc01d1bc47$7600f850$6202e8f0$@freeswitch.org>
Message-ID: <CAC8JMOJLH_b1V+7RGumuaq9c38O3rm8aWzP_NP6hW5ii9cOFXQ@mail.gmail.com>

Yep, thanks for the context Ken!

On Wed, Jun 1, 2016 at 1:52 PM, Ken Rice <krice at freeswitch.org> wrote:

> Keep in mind that with SIP if the server is listening you are supposed to
> respond. Not responding is a violation of the RFC? now that being side?
> using DPI via IPTables is a perfect way to dissuade the scanners? and btw,
> if you are using TLS, they don?t even need to see SIP to know you have
> something listening on TCP on the SIP port now, you?re syn-ack in reply to
> their syn already told them that?.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Oleg Stolyar
> *Sent:* Wednesday, June 1, 2016 3:39 PM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Is there a way for FS not to send back
> any response to INVITE
>
>
>
> Thanks Jurijs!
>
>
>
> Unfortunately we do need to use TLS.
>
>
>
> On Wed, Jun 1, 2016 at 1:26 PM, Jurijs Ivolga <jurijs.ivolga at gmail.com>
> wrote:
>
> Hi Oleg,
>
> With iptables you can block based on what is inside SIP packet(off cause
> if you are not using TLS), take a look on link below:
>
>
> http://www.bertera.it/index.php/2014/01/22/sip-facket-filtering-with-iptables/
>
> It is not best way to achieve what you need, cause as far as I know it is
> resource consuming operations. Best way will be to use Kamailio as SIP
> proxy in front.
>
> With kind regards,
>
>
> Jurijs
>
>
>
> On Wed, Jun 1, 2016 at 11:05 PM, Oleg Stolyar <olegstolyar at gmail.com>
> wrote:
>
> Thanks guys!  IP tables is how we block most traffic but we can only block
> traffic by port.  In this case it's about invalid INVITES coming in on a
> valid port.
>
>
>
> Do you think this functionality would be useful?
>
> Is it worth opening a feature request and perhaps putting a bounty on it?
>
> Any idea of the effort?
>
>
>
> On Wed, Jun 1, 2016 at 1:00 PM, Michael Jerris <mike at jerris.com> wrote:
>
> The only way with our current sip module to accomplish either of these
> would be to put a sip proxy out front to handle that behavior, or to
> somehow use iptables to block the traffic
>
>
>
> On Jun 1, 2016, at 3:40 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>
>
>
> Hi,
>
>
>
> In order to protect against scanning attacks I'd like for FS to not
> respond to INVITES unless they match certain conditions.
>
>
>
> I understand that currently FS always responds with 100 Trying right away
> before processing the call and then, if the call does not match anything in
> the dialplan, responds with a 302 Moved Temporarily.
>
>
>
> The 302 can be replaced with another response code (for example 403
> Forbidden which is what I am doing now) using the *respond* dialplan app.
>   However, that might encourage the scanner to keep trying.
>
>
>
> So I guess there are two questions:
>
>
>
> 1. Is there a way not to send back 100 Trying at all?
>
>
>
> 2. Is there a way to not send any final response?
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From naveen32india at gmail.com  Thu Jun  2 01:44:03 2016
From: naveen32india at gmail.com (Naveen Tamanam)
Date: Thu, 2 Jun 2016 03:14:03 +0530
Subject: [Freeswitch-users] Unable to install module mod_flite on CentOS
Message-ID: <CABsWbRbrgh3g9yS9tJF87ARg0nVv3qi=1y-FWSjaT=sa4ZSgdw@mail.gmail.com>

Hi All,

I'm trying to compile freeswitch with module mod_flite. I have downloaded
the module from
     https://freeswitch.org/stash/scm/sd/libflite.git
Compiled it and installed successfully. Then trying to compile  freeswitch
with mod_flite enabled, still I'm getting following error (for make),

making all mod_flite
make[4]: Entering directory
`/usr/local/src/freeswitch/src/mod/asr_tts/mod_flite'
Makefile:886: *** You must install libflite-dev to build mod_flite.  Stop.
make[4]: Leaving directory
`/usr/local/src/freeswitch/src/mod/asr_tts/mod_flite'
make[3]: *** [mod_flite-all] Error 1
make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod'
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory `/usr/local/src/freeswitch/src'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/usr/local/src/freeswitch'
make: *** [all] Error 2

I'm using centOS 7
freeswitch version --  1.6.2








-- 
Thanks & Regards,
Naveen Tamanam
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From max at nysolutions.com  Thu Jun  2 03:11:16 2016
From: max at nysolutions.com (Moishe Grunstein)
Date: Wed, 1 Jun 2016 23:11:16 +0000
Subject: [Freeswitch-users] Unable to install module mod_flite on CentOS
In-Reply-To: <CABsWbRbrgh3g9yS9tJF87ARg0nVv3qi=1y-FWSjaT=sa4ZSgdw@mail.gmail.com>
References: <CABsWbRbrgh3g9yS9tJF87ARg0nVv3qi=1y-FWSjaT=sa4ZSgdw@mail.gmail.com>
Message-ID: <ed34459e2c784a6dbf5d90c57f73b0e7@nysolutions.com>

Did you install libflite-dev?

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-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Naveen Tamanam
Sent: Wednesday, June 1, 2016 5:44 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Unable to install module mod_flite on CentOS

Hi All,
I'm trying to compile freeswitch with module mod_flite. I have downloaded the module from
     https://freeswitch.org/stash/scm/sd/libflite.git
Compiled it and installed successfully. Then trying to compile  freeswitch with mod_flite enabled, still I'm getting following error (for make),

making all mod_flite
make[4]: Entering directory `/usr/local/src/freeswitch/src/mod/asr_tts/mod_flite'
Makefile:886: *** You must install libflite-dev to build mod_flite.  Stop.
make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod/asr_tts/mod_flite'
make[3]: *** [mod_flite-all] Error 1
make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod'
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory `/usr/local/src/freeswitch/src'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/usr/local/src/freeswitch'
make: *** [all] Error 2
I'm using centOS 7
freeswitch version --  1.6.2






--
Thanks & Regards,
Naveen Tamanam
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From naveen32india at gmail.com  Thu Jun  2 03:19:55 2016
From: naveen32india at gmail.com (Naveen Tamanam)
Date: Thu, 2 Jun 2016 04:49:55 +0530
Subject: [Freeswitch-users] Unable to install module mod_flite on CentOS
In-Reply-To: <ed34459e2c784a6dbf5d90c57f73b0e7@nysolutions.com>
References: <CABsWbRbrgh3g9yS9tJF87ARg0nVv3qi=1y-FWSjaT=sa4ZSgdw@mail.gmail.com>
	<ed34459e2c784a6dbf5d90c57f73b0e7@nysolutions.com>
Message-ID: <CABsWbRauppe2KKQzEv92+sb8HrbQANaMVG7=AvXoDp_XnMGJ0Q@mail.gmail.com>

Yes, I have installed libflite-dev from
https://freeswitch.org/stash/scm/sd/libflite.git
Manually compiled it and installed it.

On Thu, Jun 2, 2016 at 4:41 AM, Moishe Grunstein <max at nysolutions.com>
wrote:

> Did you install libflite-dev?
>
>
>
> 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-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Naveen
> Tamanam
> *Sent:* Wednesday, June 1, 2016 5:44 PM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] Unable to install module mod_flite on CentOS
>
>
>
> Hi All,
>
> I'm trying to compile freeswitch with module mod_flite. I have downloaded
> the module from
>      https://freeswitch.org/stash/scm/sd/libflite.git
>
> Compiled it and installed successfully. Then trying to compile  freeswitch
> with mod_flite enabled, still I'm getting following error (for make),
>
> making all mod_flite
> make[4]: Entering directory
> `/usr/local/src/freeswitch/src/mod/asr_tts/mod_flite'
> Makefile:886: *** You must install libflite-dev to build mod_flite.  Stop.
> make[4]: Leaving directory
> `/usr/local/src/freeswitch/src/mod/asr_tts/mod_flite'
> make[3]: *** [mod_flite-all] Error 1
> make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod'
> make[2]: *** [all-recursive] Error 1
> make[2]: Leaving directory `/usr/local/src/freeswitch/src'
> make[1]: *** [all-recursive] Error 1
> make[1]: Leaving directory `/usr/local/src/freeswitch'
> make: *** [all] Error 2
>
> I'm using centOS 7
>
> freeswitch version --  1.6.2
>
>
>
>
>
>
>
> --
>
> Thanks & Regards,
> Naveen Tamanam
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Thanks & Regards,
Naveen Tamanam
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From daveh at beachdognet.com  Thu Jun  2 03:49:20 2016
From: daveh at beachdognet.com (Dave Horton)
Date: Wed, 1 Jun 2016 19:49:20 -0400
Subject: [Freeswitch-users] Is there a way for FS not to send back any
	response to INVITE
In-Reply-To: <mailman.2968.1464822676.8223.freeswitch-users@lists.freeswitch.org>
References: <mailman.2968.1464822676.8223.freeswitch-users@lists.freeswitch.org>
Message-ID: <219D1636-2E54-4CAF-BA72-B334498A7F83@beachdognet.com>

I do this with my proxy that fronts freeswitch (https://www.npmjs.com/package/drachtio-fs-load-balancing-proxy <https://www.npmjs.com/package/drachtio-fs-load-balancing-proxy>) by evaluating SIP headers for the common sip scanner signatures and dynamically adding the senders to an iptable black list.  This does result in a 100 Trying going back to the sender (since that is sent by my sip stack before proxy processing) but no final response, and every subsequent message from that address is silently dropped. 
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From schoch+freeswitch.org at xwin32.com  Thu Jun  2 03:51:50 2016
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Wed, 1 Jun 2016 16:51:50 -0700
Subject: [Freeswitch-users] Run a LUA script when voicemail received
In-Reply-To: <CAFmZFiN+q-MgFr6PfZEnDVe9hvqikASGwvCuS4A4orvX57RS7Q@mail.gmail.com>
References: <CAOkuKdfnwFsKN4H7ZxB+dvfGSGnAby59AM+kO4MCrLuJWa=Qag@mail.gmail.com>
	<CAEwTsAg0rPXMtvJtGns1kwcNWRT2rHJSnn+psXfnF9_BMdV3Ww@mail.gmail.com>
	<CAFmZFiNHye2Tujfwsw8mh9tc+bnZNxSbxTt670W0L3UtCKszGg@mail.gmail.com>
	<CAOkuKdczNFzL76oRF1f2SARJ7700owGmMnJOrtTZA6x=5AsLJQ@mail.gmail.com>
	<CAOkuKde_1pK6TzA4-9qKSVuh8A0s7i1qs2n4uUt-5-b8=HSgKQ@mail.gmail.com>
	<CAOkuKdcvRDVkmOhdMQ+7VB8edhdbzMX+25-wRAxgPSn-waX4iw@mail.gmail.com>
	<CAOkuKddVsmLA-+YjQo7myFF7HXta6K468t+7WooETYU6=_vJkw@mail.gmail.com>
	<CAFmZFiN+q-MgFr6PfZEnDVe9hvqikASGwvCuS4A4orvX57RS7Q@mail.gmail.com>
Message-ID: <CAOkuKdfD2ZsH-s_zNuiCmewbiOYRVOv2B_rP2eY5aTsrGwxjUA@mail.gmail.com>

I guess I'm doing it right. I've been programming in perl for the last few
months, so I'm just not used to the lua way. I'm pretty close to a
solution, but what I would like to get now is the callee number - that is,
the main number dialed before it was transferred to an extension. I tried
"show detailed_calls", but the only numbers that gave me were the caller-ID
number (which I already have from the event), and the number called (which
is only the extension, and is also already present in the event).

It's not a big deal, since I can use the outbound_caller_id_number
variable, but there are some voice mailboxes here where 2 different called
numbers end up at the same extension, and same voice box.

-- 
Steve

On Tue, May 31, 2016 at 6:46 PM, Abaci B <abaci64 at gmail.com> wrote:

> This is the correct way to get user data, why do you consider this unclean?
>
> On Tue, May 31, 2016 at 8:10 PM, Steven Schoch <
> schoch+freeswitch.org at xwin32.com> wrote:
>
>> I may have just answered my own question again. I found a way, but please
>> let me know if you know of a cleaner way:
>>
>> user = event:getHeader("VM-User")
>> domain = event:getHeader("VM-Domain")
>> api = freeswitch.API();
>> from = api:executeString("user_data " ..user .. "@" .. domain ..  " var
>> outbound_caller_id_number")
>> to = api:executeString("user_data " .. user .. "@" .. domain .. " param
>> vm-notify-sms")
>>
>> This should get me what I need to send the text.
>>
>> --
>> Steve
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From naveen32india at gmail.com  Thu Jun  2 06:51:14 2016
From: naveen32india at gmail.com (Naveen Tamanam)
Date: Thu, 2 Jun 2016 08:21:14 +0530
Subject: [Freeswitch-users] How to pause and resume the IVR flow
Message-ID: <CABsWbRbct78gcnK39GdaFgRtDFQmrubsKXnb2A66YpQi7O=azw@mail.gmail.com>

Hi All,

I have a simple question. Is there any way to pause and resume the IVR flow
in freeswiitch.

Case Study:

?Controlling call through event socket. I would like to play few sound
files from event socket by pausing the running IVR and then once I finished
playing sounds using event socket, I would like to resume the IVR flow.

Let's say IVR is running, I would like to execute random action from event
socket by PAUSING IVR, once action is finished I would to resume the IVR
from from where I paused it.

Is it possible ?

-- 
Thanks & Regards,
Naveen Tamanam
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From s.safarov at gmail.com  Thu Jun  2 09:38:55 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 02 Jun 2016 05:38:55 +0000
Subject: [Freeswitch-users] Unable to install module mod_flite on CentOS
In-Reply-To: <CABsWbRauppe2KKQzEv92+sb8HrbQANaMVG7=AvXoDp_XnMGJ0Q@mail.gmail.com>
References: <CABsWbRbrgh3g9yS9tJF87ARg0nVv3qi=1y-FWSjaT=sa4ZSgdw@mail.gmail.com>
	<ed34459e2c784a6dbf5d90c57f73b0e7@nysolutions.com>
	<CABsWbRauppe2KKQzEv92+sb8HrbQANaMVG7=AvXoDp_XnMGJ0Q@mail.gmail.com>
Message-ID: <CAHtxdDds+72m1ZWu9jNBha1j_nKhN+9iq3CSQgziY=wpbdOrbA@mail.gmail.com>

Check flite version. When package is old, then FS says "You must install
libflite-dev to build mod_flite".
Check configure script log.

Sergey

??, 2 ???. 2016 ?. ? 2:20, Naveen Tamanam <naveen32india at gmail.com>:

> Yes, I have installed libflite-dev from
> https://freeswitch.org/stash/scm/sd/libflite.git
> Manually compiled it and installed it.
>
> On Thu, Jun 2, 2016 at 4:41 AM, Moishe Grunstein <max at nysolutions.com>
> wrote:
>
>> Did you install libflite-dev?
>>
>>
>>
>> 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-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Naveen
>> Tamanam
>> *Sent:* Wednesday, June 1, 2016 5:44 PM
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* [Freeswitch-users] Unable to install module mod_flite on
>> CentOS
>>
>>
>>
>> Hi All,
>>
>> I'm trying to compile freeswitch with module mod_flite. I have downloaded
>> the module from
>>      https://freeswitch.org/stash/scm/sd/libflite.git
>>
>> Compiled it and installed successfully. Then trying to compile
>> freeswitch with mod_flite enabled, still I'm getting following error (for
>> make),
>>
>> making all mod_flite
>> make[4]: Entering directory
>> `/usr/local/src/freeswitch/src/mod/asr_tts/mod_flite'
>> Makefile:886: *** You must install libflite-dev to build mod_flite.  Stop.
>> make[4]: Leaving directory
>> `/usr/local/src/freeswitch/src/mod/asr_tts/mod_flite'
>> make[3]: *** [mod_flite-all] Error 1
>> make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod'
>> make[2]: *** [all-recursive] Error 1
>> make[2]: Leaving directory `/usr/local/src/freeswitch/src'
>> make[1]: *** [all-recursive] Error 1
>> make[1]: Leaving directory `/usr/local/src/freeswitch'
>> make: *** [all] Error 2
>>
>> I'm using centOS 7
>>
>> freeswitch version --  1.6.2
>>
>>
>>
>>
>>
>>
>>
>> --
>>
>> Thanks & Regards,
>> Naveen Tamanam
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Thanks & Regards,
> Naveen Tamanam
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From jleung at v10networks.ca  Thu Jun  2 10:32:36 2016
From: jleung at v10networks.ca (Jeff Leung)
Date: Wed, 1 Jun 2016 23:32:36 -0700
Subject: [Freeswitch-users] Unable to install module mod_flite on CentOS
In-Reply-To: <CAHtxdDds+72m1ZWu9jNBha1j_nKhN+9iq3CSQgziY=wpbdOrbA@mail.gmail.com>
References: <CABsWbRbrgh3g9yS9tJF87ARg0nVv3qi=1y-FWSjaT=sa4ZSgdw@mail.gmail.com><ed34459e2c784a6dbf5d90c57f73b0e7@nysolutions.com><CABsWbRauppe2KKQzEv92+sb8HrbQANaMVG7=AvXoDp_XnMGJ0Q@mail.gmail.com>
	<CAHtxdDds+72m1ZWu9jNBha1j_nKhN+9iq3CSQgziY=wpbdOrbA@mail.gmail.com>
Message-ID: <A70AC0C1B7E7FD478A6BF19705461AA0C966@IMBER.smb.curriegrad2004.ca>

> Check flite version. When package is old, then FS says "You must install
> libflite-dev to build mod_flite".
> Check configure script log.
The way how FreeSWITCH checks for flite is it uses pkg-config. If there's a pkg-config .pc file for flite, then it'll detect it, otherwise it'll say the flite libs aren't installed but in reality it was installed.

The libflite package that came with Ubuntu 16.04 and Debian 8 didn't have the .pc file for libflite, so the developers had to bring in a version that had that necessary .pc file

> Sergey
> 
> ??, 2 ???. 2016 ?. ? 2:20, Naveen Tamanam <naveen32india at gmail.com>:
> 
> 
> 	Yes, I have installed libflite-dev from
> https://freeswitch.org/stash/scm/sd/libflite.git
> 
> 	Manually compiled it and installed it.
> 
> 
> 	On Thu, Jun 2, 2016 at 4:41 AM, Moishe Grunstein
> <max at nysolutions.com> wrote:
> 
> 
> 		Did you install libflite-dev?
> 
> 
> 
> 		Thanks,
> 
> 
> 
> 		Moishe Grunstein
> 
> 		Tornado Computer Systems, Inc.
> 
> 		212.400.7650 888.IPPBX.US
> 		Service Request Email: 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-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Naveen Tamanam
> 		Sent: Wednesday, June 1, 2016 5:44 PM
> 		To: FreeSWITCH Users Help <freeswitch-
> users at lists.freeswitch.org>
> 		Subject: [Freeswitch-users] Unable to install module
> mod_flite on CentOS
> 
> 
> 
> 		Hi All,
> 
> 		I'm trying to compile freeswitch with module mod_flite. I
> have downloaded the module from
> 		     https://freeswitch.org/stash/scm/sd/libflite.git
> 
> 		Compiled it and installed successfully. Then trying to compile
> freeswitch with mod_flite enabled, still I'm getting following error (for
> make),
> 
> 		making all mod_flite
> 		make[4]: Entering directory
> `/usr/local/src/freeswitch/src/mod/asr_tts/mod_flite'
> 		Makefile:886: *** You must install libflite-dev to build
> mod_flite.  Stop.
> 		make[4]: Leaving directory
> `/usr/local/src/freeswitch/src/mod/asr_tts/mod_flite'
> 		make[3]: *** [mod_flite-all] Error 1
> 		make[3]: Leaving directory
> `/usr/local/src/freeswitch/src/mod'
> 		make[2]: *** [all-recursive] Error 1
> 		make[2]: Leaving directory `/usr/local/src/freeswitch/src'
> 		make[1]: *** [all-recursive] Error 1
> 		make[1]: Leaving directory `/usr/local/src/freeswitch'
> 		make: *** [all] Error 2
> 
> 		I'm using centOS 7
> 
> 		freeswitch version --  1.6.2
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 		--
> 
> 		Thanks & Regards,
> 		Naveen Tamanam
> 
> 
> 
> 	____________________________________________________
> _____________________
> 		Professional FreeSWITCH Consulting Services:
> 		consulting at freeswitch.org
> 		http://www.freeswitchsolutions.com
> 
> 		Official FreeSWITCH Sites
> 		http://www.freeswitch.org
> 		http://confluence.freeswitch.org
> 		http://www.cluecon.com
> 
> 		FreeSWITCH-users mailing list
> 		FreeSWITCH-users at lists.freeswitch.org
> 		http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> 	UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitc
> h-users
> 		http://www.freeswitch.org
> 
> 
> 
> 
> 
> 	--
> 
> 	Thanks & Regards,
> 	Naveen Tamanam
> 
> 	____________________________________________________
> _____________________
> 	Professional FreeSWITCH Consulting Services:
> 	consulting at freeswitch.org
> 	http://www.freeswitchsolutions.com
> 
> 	Official FreeSWITCH Sites
> 	http://www.freeswitch.org
> 	http://confluence.freeswitch.org
> 	http://www.cluecon.com
> 
> 	FreeSWITCH-users mailing list
> 	FreeSWITCH-users at lists.freeswitch.org
> 	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 	UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitc
> h-users
> 	http://www.freeswitch.org


From ssinyagin at gmail.com  Thu Jun  2 16:08:41 2016
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Thu, 2 Jun 2016 14:08:41 +0200
Subject: [Freeswitch-users] How to pause and resume the IVR flow
In-Reply-To: <CABsWbRbct78gcnK39GdaFgRtDFQmrubsKXnb2A66YpQi7O=azw@mail.gmail.com>
References: <CABsWbRbct78gcnK39GdaFgRtDFQmrubsKXnb2A66YpQi7O=azw@mail.gmail.com>
Message-ID: <CACOYK=jtgwmTqgaD9QG_4qt8WvZAT8YbMNas5NP21pg_ho_1eg@mail.gmail.com>

why not implementing the whole IVR via ESL connection? Then you
control every aspect of the workflow.

In regards to your question, I'm not sure, needs testing. I think it
should be to gain control of a running session, but I'm not sure you
can easily resume the IVR operation after that.



On Thu, Jun 2, 2016 at 4:51 AM, Naveen Tamanam <naveen32india at gmail.com> wrote:
> Hi All,
>
> I have a simple question. Is there any way to pause and resume the IVR flow
> in freeswiitch.
>
> Case Study:
>
> Controlling call through event socket. I would like to play few sound files
> from event socket by pausing the running IVR and then once I finished
> playing sounds using event socket, I would like to resume the IVR flow.
>
> Let's say IVR is running, I would like to execute random action from event
> socket by PAUSING IVR, once action is finished I would to resume the IVR
> from from where I paused it.
>
> Is it possible ?
>
> --
> Thanks & Regards,
> Naveen Tamanam
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From william at magicwilly.info  Thu Jun  2 03:52:11 2016
From: william at magicwilly.info (William Scott)
Date: Thu, 2 Jun 2016 09:52:11 +1000
Subject: [Freeswitch-users] SwitchPi
In-Reply-To: <00bb01d1bc20$ab399cb0$01acd610$@freeswitch.org>
References: <CABV9BUu8ygpbbGLzxmzG70NsiS7dj5D4WRkNaQH6Hzex_Q38yg@mail.gmail.com>
	<00bb01d1bc20$ab399cb0$01acd610$@freeswitch.org>
Message-ID: <CABV9BUuns9Z+-KYJvj8LeRrDUkKBnL-Dh46dXJ-Y2kDFent2Rg@mail.gmail.com>

Hi

Yes, I have FusionPBX/freeSwitch running on a RPi3. Will the ARM build ever
become "Official"?

I've watched a 'tube video or two. What is the functionality of the custom
boards?
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From mike at jerris.com  Thu Jun  2 20:49:52 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 2 Jun 2016 12:49:52 -0400
Subject: [Freeswitch-users] Unable to install module mod_flite on CentOS
In-Reply-To: <CABsWbRauppe2KKQzEv92+sb8HrbQANaMVG7=AvXoDp_XnMGJ0Q@mail.gmail.com>
References: <CABsWbRbrgh3g9yS9tJF87ARg0nVv3qi=1y-FWSjaT=sa4ZSgdw@mail.gmail.com>
	<ed34459e2c784a6dbf5d90c57f73b0e7@nysolutions.com>
	<CABsWbRauppe2KKQzEv92+sb8HrbQANaMVG7=AvXoDp_XnMGJ0Q@mail.gmail.com>
Message-ID: <0406510B-6297-4834-8030-B4E4B7B5278E@jerris.com>

This shoud be the right one.  Did you run freeswitch configure after you installed this?

> On Jun 1, 2016, at 7:19 PM, Naveen Tamanam <naveen32india at gmail.com> wrote:
> 
> Yes, I have installed libflite-dev from  https://freeswitch.org/stash/scm/sd/libflite.git <https://freeswitch.org/stash/scm/sd/libflite.git>
> Manually compiled it and installed it.
> 
> On Thu, Jun 2, 2016 at 4:41 AM, Moishe Grunstein <max at nysolutions.com <mailto:max at nysolutions.com>> wrote:
> Did you install libflite-dev?
> 
>  
> 
> Thanks,
> 
>  
> 
> Moishe Grunstein
> 
> Tornado Computer Systems, Inc.
> 
> 212.400.7650 888.IPPBX.US <http://888.ippbx.us/>
> Service Request Email: support at nysolutions.com <mailto:support at nysolutions.com>
> <image001.jpg> <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-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Naveen Tamanam
> Sent: Wednesday, June 1, 2016 5:44 PM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> Subject: [Freeswitch-users] Unable to install module mod_flite on CentOS
> 
>  
> 
> Hi All,
> 
> I'm trying to compile freeswitch with module mod_flite. I have downloaded the module from   
>      https://freeswitch.org/stash/scm/sd/libflite.git <https://freeswitch.org/stash/scm/sd/libflite.git>
> Compiled it and installed successfully. Then trying to compile  freeswitch with mod_flite enabled, still I'm getting following error (for make), 
> 
> making all mod_flite
> make[4]: Entering directory `/usr/local/src/freeswitch/src/mod/asr_tts/mod_flite'
> Makefile:886: *** You must install libflite-dev to build mod_flite.  Stop.
> make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod/asr_tts/mod_flite'
> make[3]: *** [mod_flite-all] Error 1
> make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod'
> make[2]: *** [all-recursive] Error 1
> make[2]: Leaving directory `/usr/local/src/freeswitch/src'
> make[1]: *** [all-recursive] Error 1
> make[1]: Leaving directory `/usr/local/src/freeswitch'
> make: *** [all] Error 2
> 
> I'm using centOS 7
> 
> freeswitch version --  1.6.2 
> 
> 
> 
> 

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From krice at freeswitch.org  Thu Jun  2 21:11:07 2016
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 2 Jun 2016 12:11:07 -0500
Subject: [Freeswitch-users] SwitchPi
In-Reply-To: <CABV9BUuns9Z+-KYJvj8LeRrDUkKBnL-Dh46dXJ-Y2kDFent2Rg@mail.gmail.com>
References: <CABV9BUu8ygpbbGLzxmzG70NsiS7dj5D4WRkNaQH6Hzex_Q38yg@mail.gmail.com>	<00bb01d1bc20$ab399cb0$01acd610$@freeswitch.org>
	<CABV9BUuns9Z+-KYJvj8LeRrDUkKBnL-Dh46dXJ-Y2kDFent2Rg@mail.gmail.com>
Message-ID: <04ea01d1bcf1$c1a373f0$44ea5bd0$@freeswitch.org>

The custom boards allow for hardware DTMF keypad interface, hook switch and a few other things like LCD for CLID on old school phones with that provision.

 

There will be official Raspian packages sooner or later? theres a know issue with the cowbuilder that?s blocking this at this time, we?re working on a work around? 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William Scott
Sent: Wednesday, June 1, 2016 6:52 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] SwitchPi

 

Hi

 

Yes, I have FusionPBX/freeSwitch running on a RPi3. Will the ARM build ever become "Official"?

 

I've watched a 'tube video or two. What is the functionality of the custom boards?



  <https://mailtrack.io/trace/mail/d1a10f8bcbe53cd87724e18dfebb1cfaa57d5246383145.png> 

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From chad at apartmentlines.com  Thu Jun  2 22:08:08 2016
From: chad at apartmentlines.com (Chad Phillips)
Date: Thu, 2 Jun 2016 13:08:08 -0500
Subject: [Freeswitch-users] What mod_verto implementations are out in the
	wild?
Message-ID: <CAJ0sghRiRmsCzx0Vm2W3EKG6T_b7a_29YZ=RH+RkmvpSTp5u+w@mail.gmail.com>

Curious if anyone knows of other well-developed mod_verto implementations
besides Verto Communicator. Documentation is still fairly thin, and it's
capable of a lot, so I'm wondering who else has climbed the mountain. :)

I have a fairly robust implementation running on
https://connect.circleanywhere.com for our 'main room' videoconference.

Not sure how useful a list of implementations would be for others, perhaps
we could add it to the mod_verto wiki page?

Chad
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From lists at kavun.ch  Fri Jun  3 00:09:29 2016
From: lists at kavun.ch (Emrah)
Date: Thu, 2 Jun 2016 22:09:29 +0200
Subject: [Freeswitch-users] Ringback after pre_answer without bridge?
In-Reply-To: <CA+GmxbYKnkDm72-0Rmmag0OGzwgm5_L=p_5wStHmrwV8n0pVcw@mail.gmail.com>
References: <CA+GmxbbJDyCmiM9LhGEBq1NLVAPwVT4oXCV=_Dpt5FsiOsvkgw@mail.gmail.com>
	<CA+GmxbYKnkDm72-0Rmmag0OGzwgm5_L=p_5wStHmrwV8n0pVcw@mail.gmail.com>
Message-ID: <3BB49779-BFA8-424E-948E-C258226315DD@kavun.ch>

I had the exact same question a few months ago. It has to be scripted and done via ESL as it is not supported natively by FS. If it can help, you can always bridge to a loopback/wait channel. 
In my scenario I wanted FS to continue dialplan execution while a ringback is played to the caller.

E
> On May 31, 2016, at 3:49 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> 
> Maybe there is also possibility  to start background MOH when there is
> no media on channel?
> 
> 2016-05-31 14:57 GMT+03:00 Dmitry Sytchev <kbdfck at gmail.com <mailto:kbdfck at gmail.com>>:
>> Hi all!
>> 
>> I'd like to resolve some unclear ringback behavior in early media
>> mode. I'm trying to implement ringback while no bridge active.
>> A calls FS, FS does something, then searching for B and bridges A to B.
>> 
>> I'd like to start playing custom ringback immediately after call from
>> A gets into FS, not when we start bridge to B.
>> 
>> I'm doing something like that:
>> 
>> <action application="set" data="ringback=%(2000, 4000, 440.0, 480.0)"/>
>> <action application="set" data="instant_ringback=true"/>
>> <action application="set" data="bridge_early_media=true"/>
>> <action application="pre_answer"/>
>> <action application="sleep" data="10000"/>
>> ...
>> <action application="bridge" data="some bridge data"/>
>> 
>> There is no ringback to A in such scenario until bridge is called in
>> further dialplan processing. I've tried all combinations of
>> ignore/bridge early media, but with no luck. Pre_answer can be called
>> directly or by FS when call is recorded, with no difference. There is
>> a wiki page that recommends usage of ring_ready:
>> 
>> https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+ring_ready
>> 
>> ring_ready works, but it can give only Ringing to calling side, so  I
>> can't pass custom RBT to calling party. It would be enough for me, but
>> once somebody calls pre_answer, no early media is passed until bridge
>> attempt.
>> 
>> What can be done to resolve this? Thanks in advance.
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> --
>> Best regards,
>> 
>> Dmitry Sytchev,
>> IT Engineer
> 
> 
> 
> -- 
> Best regards,
> 
> Dmitry Sytchev,
> IT Engineer
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From bipin at xbipin.com  Fri Jun  3 00:57:56 2016
From: bipin at xbipin.com (Bipin Patel)
Date: Fri, 03 Jun 2016 00:57:56 +0400
Subject: [Freeswitch-users] VC codecs
Message-ID: <15512e88038.2787.b07ebdf329620b8089087c7205b03f01@xbipin.com>

I wanted to know what codecs can I use with verto communicator for audio 
only and to bridge calls to sip gateway. Basically wanted to know if it can 
handle g711u because most of my gateways use that only along with g729 and 
g723 but am not interested in those





From mike at jerris.com  Fri Jun  3 01:11:11 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 2 Jun 2016 17:11:11 -0400
Subject: [Freeswitch-users] VC codecs
In-Reply-To: <15512e88038.2787.b07ebdf329620b8089087c7205b03f01@xbipin.com>
References: <15512e88038.2787.b07ebdf329620b8089087c7205b03f01@xbipin.com>
Message-ID: <2DB28A09-6E90-4CAD-9D1C-D84ED68AAAB0@jerris.com>

VC doesn't limit the codecs... the end devices and freeswitch codec support would.  i have seen g711 available on all the werbtc implementations i've seen.


> On Jun 2, 2016, at 4:57 PM, Bipin Patel <bipin at xbipin.com> wrote:
> 
> I wanted to know what codecs can I use with verto communicator for audio 
> only and to bridge calls to sip gateway. Basically wanted to know if it can 
> handle g711u because most of my gateways use that only along with g729 and 
> g723 but am not interested in those
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From schoch+freeswitch.org at xwin32.com  Fri Jun  3 02:26:00 2016
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Thu, 2 Jun 2016 15:26:00 -0700
Subject: [Freeswitch-users] Run a LUA script when voicemail received
In-Reply-To: <CAFmZFiN+q-MgFr6PfZEnDVe9hvqikASGwvCuS4A4orvX57RS7Q@mail.gmail.com>
References: <CAOkuKdfnwFsKN4H7ZxB+dvfGSGnAby59AM+kO4MCrLuJWa=Qag@mail.gmail.com>
	<CAEwTsAg0rPXMtvJtGns1kwcNWRT2rHJSnn+psXfnF9_BMdV3Ww@mail.gmail.com>
	<CAFmZFiNHye2Tujfwsw8mh9tc+bnZNxSbxTt670W0L3UtCKszGg@mail.gmail.com>
	<CAOkuKdczNFzL76oRF1f2SARJ7700owGmMnJOrtTZA6x=5AsLJQ@mail.gmail.com>
	<CAOkuKde_1pK6TzA4-9qKSVuh8A0s7i1qs2n4uUt-5-b8=HSgKQ@mail.gmail.com>
	<CAOkuKdcvRDVkmOhdMQ+7VB8edhdbzMX+25-wRAxgPSn-waX4iw@mail.gmail.com>
	<CAOkuKddVsmLA-+YjQo7myFF7HXta6K468t+7WooETYU6=_vJkw@mail.gmail.com>
	<CAFmZFiN+q-MgFr6PfZEnDVe9hvqikASGwvCuS4A4orvX57RS7Q@mail.gmail.com>
Message-ID: <CAOkuKdf_u0h8pOQSJBBntFNaKr0OeFseyEmTOuDiBrJZpimt0A@mail.gmail.com>

I got it working! The goal was to run a script to inform users via a text
message when they get a voicemail.
This script looks for a "vm-notify-sms" param in the user's directory
information. It sends the text from user's outbound caller-ID number, which
is set in the standard FS configuration. I'm using the SMS API from
Flowroute.com.

--[[ This script is called by configuration in
autoload_configs/lua.conf.xml.
     It is called on a "vm:maintenance" event.
]]

-- The following is specifc to the Flowroute SMS API,
-- documented at https://developer.flowroute.com/docs/messaging
url = "https://api.flowroute.com/v2/messages"
authentication = "user:password"

action = event:getHeader("VM-Action")
if action ~= "leave-message" then return end
user = event:getHeader("VM-User")
domain = event:getHeader("VM-Domain")
caller_id_name = event:getHeader("VM-Caller-ID-Name")
caller_id_number = event:getHeader("VM-Caller-ID-Number")
--[[ We don't care about these:
file_path = event:getHeader("VM-File-Path")
flags = event:getHeader("VM-Flags")
folder = event:getHeader("VM-Folder")
uuid = event:getHeader("VM-UUID")
--]]
message_len = event:getHeader("VM-Message-Len")
timestamp = event:getHeader("VM-Timestamp")
api = freeswitch.API();
from = api:execute("user_data", user .. "@" .. domain ..  " var
outbound_caller_id_number")
to = api:execute("user_data", user .. "@" .. domain .. " param
vm-notify-sms")
if to == "" then return end

if caller_id_name == ''
        or caller_id_name == 'UNKNOWN'
        or caller_id_name == 'UNASSIGNED'
        or caller_id_name == 'WIRELESS CALLER'
        or caller_id_name == 'TOLL FREE CALL'
        or caller_id_name == 'Anonymous'
        or caller_id_name == 'Unavailable'
        then caller_id_name = nil end
if caller_id_number == ''
        then caller_id_number = nil end
message = "Voicemail "
if caller_id_name
        then message = message .. "from " .. caller_id_name .. " (" ..
 caller_id_number .. ")"
        elseif caller_id_number
        then message = message .. "from " .. caller_id_number end
message = message .. " at " .. os.date("%I:%M %p", timestamp)
message = message .. " length " .. message_len .. " seconds"
message = message .. " to box " .. user
message = message .. "."
-- Blindly send the text
data = '{ "to": "' .. to .. '", "from": "' .. from .. '", "body": "' ..
message .. '"}'
--[[
     Until either Freeswitch has the ability to pass authentication data
     to the mod_curl API (FS-9223), or the Flowroute API has the ability
     to take authentication data in the POST body (they are working on it),
     we are forced to spawn a shell to use the curl from our OS.
]]
-- api.execute("curl", url .. " auth " .. authentication .. " post " ..
data)
os.execute("curl -u " .. authentication .. " -H 'Content-Type:
application/json' -X POST -d '" .. data .. "' "  .. url)
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From cody.rosenbloom at gmail.com  Fri Jun  3 02:33:25 2016
From: cody.rosenbloom at gmail.com (Cody Rosenbloom)
Date: Thu, 2 Jun 2016 18:33:25 -0400
Subject: [Freeswitch-users] Lua Script Does Not Get Executed
Message-ID: <CAPrxTnpVFh8fzaqPzMKykdq5f7s2Pvktg2amNMSzS3tyw7JChg@mail.gmail.com>

Hi,

I am very new to freeswitch and Lua. I'm trying to get a lua script which
I've set up to send voicemails but as far as I can tell, the script is
never even executed. This is the basic sequence of events.

-> Caller leaves a voicemail. This is stored on the server, no problem.
->After voicemail is stored, I call a lua script passing the email id of
the user and his extension
->The os.execute command within the script passes these values to a simple
shell script which grabs the voicemail audio file, logs some debug output
to an output file so I can verify its been run and then sends me an email.
When I test the script from fs_cli console using

lua <scriptName> <args> , it works fine and I see the debug output logged.
When I actually call and leave a voicemail, this does not happen. Can
someone point out what I'm doing wrong ?

My xml and lua script are pasted at http://pastebin.com/2N6LcfuV

Thank you very much!

Cody
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From abaci64 at gmail.com  Fri Jun  3 02:54:53 2016
From: abaci64 at gmail.com (Abaci B)
Date: Thu, 2 Jun 2016 18:54:53 -0400
Subject: [Freeswitch-users] Windows build
In-Reply-To: <CAEwTsAgU8dn+BqO0K94PPRvqTOKDZFBHdgw51tmkfKSrtmzbXA@mail.gmail.com>
References: <CAC7ttFFYf+dt3NfgRQy0R5PbNut0=-=mmxv6+gTvUYFB1RsKBA@mail.gmail.com>
	<CAPEdYThObr1ha7ffrFNijAj3H30Z6Zk41TETEgdoeZWxLu6igg@mail.gmail.com>
	<CAC7ttFE98byMZX+uNYUcLJ-A8+WnQwcwXBqqPO-g2fBu3XRfvA@mail.gmail.com>
	<CAPEdYTghm8J20E5COERvG1TidbYNUqcVsjD1TngTyH-zwWE52w@mail.gmail.com>
	<CAC7ttFGRJQAFn0vC3sCC1tguvWmZ==BpKR4i-=UWjDPRgGx-hA@mail.gmail.com>
	<9589f6e5d757412ba5d620cf8e2bf26c@imladris.sermotec.local>
	<56E10D5D.1000303@zg.t-com.hr>
	<38494249596647c6a08591eca41e3000@imladris.sermotec.local>
	<CAC7ttFFbv9YAxXNg7zjc1B9cxLS74v6f6HfBpSt09sULor6EoA@mail.gmail.com>
	<CAHtxdDdY2JRqVuw-W0b+UF0Z_ZUGbCj_iUFUOXi=nrxDDzKb9A@mail.gmail.com>
	<CAC7ttFErJO-gTLBQTHy_6zCcRWBxOYgsXtJXqMC0U8Tv+79AOg@mail.gmail.com>
	<3ED4D03E-7C8C-4DE6-9DC1-1C6E039C5355@jerris.com>
	<CAC7ttFFfGFD+mzKRZWvwNy3BYoKK7FXY1pBgPGUMbj79qYeMmA@mail.gmail.com>
	<CAEwTsAgU8dn+BqO0K94PPRvqTOKDZFBHdgw51tmkfKSrtmzbXA@mail.gmail.com>
Message-ID: <CAFmZFiPE_9gHbwquw=9-k2WbJMs1CBDV3hmRodX60dKmWJFUbA@mail.gmail.com>

any updates on getting updated windows binaries, or is there another place
to download a more up to date version?

On Wed, Apr 6, 2016 at 12:01 PM, Michael Jerris <mike at jerris.com> wrote:

> I'll work on getting new ones out there, but it's easy to build it yourself
>
>
> On Wednesday, April 6, 2016, Gregor Nanger <gregor at infomedia.si> wrote:
>
>> Thank you Michael!
>>
>> Maybe it is something with my configuration of VS...
>>
>> Can you copy your build also on
>> http://files.freeswitch.org/windows/installer/x64/?
>>
>> 2016-04-06 1:08 GMT+02:00 Michael Jerris <mike at jerris.com>:
>>
>>> I just built windows fine yesterday.  It may be unhappy with rebuild,
>>> but I built fine. (this should cover master).... mod_avmd might have been
>>> failing for a few commits, but I pushed the fix for that module yesterday,
>>> all the others in default configuration built fine.
>>>
>>> On Apr 5, 2016, at 6:38 PM, Gregor Nanger <gregor at infomedia.si> wrote:
>>>
>>> Just bumping this post if someone has any hints.
>>>
>>> I am trying to build Freeswitch with Visual studio 2015, but keep
>>> getting error:
>>> Error MSB4057 The target "v8:Rebuild" does not exist in the project.
>>> [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln] libv8
>>> ? ?
>>> C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
>>> ?"
>>>
>>>
>>> Would realy like to successfuly build Freeswitch in windows. I tried
>>> with 1.6 branch.
>>>
>>> Best regards, Gregor
>>>
>>> 2016-03-17 5:43 GMT+01:00 Sergey Safarov <s.safarov at gmail.com>:
>>>
>>>> One week ago I successfully compiled mod_V8 on CentOS 7.
>>>> May be switch to Linux?
>>>>
>>>> On Thu, Mar 17, 2016 at 12:49 AM, Gregor Nanger <gregor at infomedia.si>
>>>> wrote:
>>>>
>>>>> Thank you Harald.
>>>>>
>>>>> I tried with latest branch 1.7 and x64 and got errors regarding
>>>>> mod_V8. I tried what Peter suggested, but I am more in c# and web projects
>>>>> and do not have experience in building  C++ projects. There is already
>>>>> prebuilt setup on freeswitch site, so someone successfully build it :-)) It
>>>>> is not so important for us at this point to make own build, so will try
>>>>> again later.
>>>>>
>>>>>
>>>>> 2016-03-11 7:32 GMT+01:00 Harald Petrovitsch <
>>>>> Harald.Petrovitsch at sermotec.at>:
>>>>>
>>>>>> Hi Gregor,
>>>>>>
>>>>>>
>>>>>>
>>>>>> I only do a
>>>>>>
>>>>>> Git.exe clone ?bv1.6
>>>>>> https://freeswitch.org/stash/scm/fs/freeswitch.git .
>>>>>>
>>>>>>
>>>>>>
>>>>>> Loaded the solution into vs2015, set configuration Win32 / Release
>>>>>> and press f7 (need to do it two times)
>>>>>>
>>>>>>
>>>>>>
>>>>>> The build ended with
>>>>>>
>>>>>> ========== Build: 20 succeeded, 0 failed, 157 up-to-date, 15 skipped
>>>>>> ==========
>>>>>>
>>>>>>
>>>>>>
>>>>>> I?ve attached a list of the generated mod folder
>>>>>>
>>>>>>
>>>>>>
>>>>>> Regards
>>>>>>
>>>>>> Harald
>>>>>>
>>>>>>
>>>>>>
>>>>>> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>>>>> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *
>>>>>> Shishko
>>>>>> *Gesendet:* Donnerstag, 10. M?rz 2016 07:00
>>>>>> *An:* freeswitch-users at lists.freeswitch.org
>>>>>>
>>>>>> *Betreff:* Re: [Freeswitch-users] Windows build
>>>>>>
>>>>>>
>>>>>>
>>>>>> Hi Harald,
>>>>>>
>>>>>> what did you do to build libv8 and mod_v8? I tried with VS2015 Update
>>>>>> 1, branch 1.6, but to no avail.
>>>>>>
>>>>>> Thanks
>>>>>>
>>>>>> On 03/07/2016 08:20 AM, Harald Petrovitsch wrote:
>>>>>>
>>>>>> Hi Gregor !
>>>>>>
>>>>>>
>>>>>>
>>>>>> V8 libs and mod builds fine here (visual Studio 2015 Sp1, 1.6 branch,
>>>>>> used tortoiseGit to download it))
>>>>>>
>>>>>>
>>>>>>
>>>>>> Regards
>>>>>>
>>>>>> Harald
>>>>>>
>>>>>>
>>>>>>
>>>>>> *Von:* freeswitch-users-bounces at lists.freeswitch.org [
>>>>>> mailto:freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag v**on
>>>>>> *Gregor Nanger
>>>>>> *Gesendet:* Montag, 07. M?rz 2016 00:19
>>>>>> *An:* FreeSWITCH Users Help
>>>>>> *Betreff:* Re: [Freeswitch-users] Windows build
>>>>>>
>>>>>>
>>>>>>
>>>>>> Thank you, H
>>>>>>
>>>>>> ?a?
>>>>>>
>>>>>> rald. This works: "AFAIK, for the ?'lame/lame.h'? you have to change
>>>>>> the include line to  only ?lame.h?"
>>>>>>
>>>>>> B
>>>>>>
>>>>>> ?ut for v8 stil do  not have solution. I do not want to exclude
>>>>>> mod_v8, since this module runs javascript. But, can you please confirm me
>>>>>> that is not yet compatible, to stop trying to solve it.
>>>>>>
>>>>>>
>>>>>> Any other suggestion what does this mean:
>>>>>> "
>>>>>> Error MSB4057 The target "v8:Rebuild" does not exist in the project.
>>>>>> [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln] libv8
>>>>>>
>>>>>> ? ?
>>>>>>
>>>>>> C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
>>>>>>
>>>>>> ?"
>>>>>>
>>>>>>
>>>>>>
>>>>>> Best regards, Gregor?
>>>>>>
>>>>>>
>>>>>>
>>>>>> 2016-03-06 16:39 GMT+01:00 Peter Olsson <peter at olssononline.se>:
>>>>>>
>>>>>> Remove mod_v8 from the build. I don't think it's compatible with
>>>>>> VS2015 for now. However, all other modules should be ok.
>>>>>>
>>>>>>
>>>>>>
>>>>>> /Peter
>>>>>>
>>>>>>
>>>>>>
>>>>>> 2016-03-06 12:31 GMT+01:00 Gregor Nanger <gregor at infomedia.si>:
>>>>>>
>>>>>> ?This helped a lot, thank you. Now I have only few errors. Any hint?
>>>>>>
>>>>>>
>>>>>>
>>>>>> Error MSB4057 The target "v8:Rebuild" does not exist in the project.
>>>>>> [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln]
>>>>>> libv8 C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
>>>>>>
>>>>>> Error MSB4057 The target "v8:Rebuild" does not exist in the project.
>>>>>> [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln] libv8
>>>>>> C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
>>>>>>
>>>>>> Error C1083 Cannot open include file: 'lame/lame.h': No such file or
>>>>>> directory mod_shout C:\Git\freeswitch\src\mod\formats\mod_shout\mod_shout.c
>>>>>> 38
>>>>>>
>>>>>> Error LNK1181 cannot open input file 'icui18n.lib' mod_v8
>>>>>> C:\Git\freeswitch\src\mod\languages\mod_v8\LINK 1
>>>>>>
>>>>>>
>>>>>>
>>>>>> ?
>>>>>>
>>>>>>
>>>>>>
>>>>>> 2016-03-06 7:53 GMT+01:00 Peter Olsson <peter at olssononline.se>:
>>>>>>
>>>>>> One common mistake is that you allow Git to modify line endings. Make
>>>>>> sure autocrlf is turned off - then clone the repository again from scratch.
>>>>>>
>>>>>>
>>>>>>
>>>>>> Also, I'm not sure if it will work in VS2015, but give it a try.
>>>>>>
>>>>>>
>>>>>>
>>>>>> /Peter
>>>>>>
>>>>>>
>>>>>>
>>>>>> 2016-03-06 2:06 GMT+01:00 Gregor Nanger <gregor at infomedia.si>:
>>>>>>
>>>>>> Hi!
>>>>>>
>>>>>>
>>>>>>
>>>>>> I want to build Freeswitch on windows with visual studio 2015.
>>>>>>
>>>>>>
>>>>>>
>>>>>> Where should I start if I get 600 errors when try to Rebuild All. I
>>>>>> opened solution and start Rebuild All, but I get so many errors that I
>>>>>> belive that I am doing something wrong.
>>>>>>
>>>>>>
>>>>>>
>>>>>> Mainl yre errors regarding:
>>>>>>
>>>>>>
>>>>>>
>>>>>> Cannot open source file....
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> Best regards, Gregor
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>>
>>>>>> *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
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>>
>>>>>> *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
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>>
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>
>>>>>> consulting at freeswitch.org
>>>>>>
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>>
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>>
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>> http://confluence.freeswitch.org
>>>>>>
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>>
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> 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
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> 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
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> 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
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gregor at infomedia.si  Fri Jun  3 03:25:08 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Fri, 3 Jun 2016 01:25:08 +0200
Subject: [Freeswitch-users] Windows build
In-Reply-To: <CAFmZFiPE_9gHbwquw=9-k2WbJMs1CBDV3hmRodX60dKmWJFUbA@mail.gmail.com>
References: <CAC7ttFFYf+dt3NfgRQy0R5PbNut0=-=mmxv6+gTvUYFB1RsKBA@mail.gmail.com>
	<CAPEdYThObr1ha7ffrFNijAj3H30Z6Zk41TETEgdoeZWxLu6igg@mail.gmail.com>
	<CAC7ttFE98byMZX+uNYUcLJ-A8+WnQwcwXBqqPO-g2fBu3XRfvA@mail.gmail.com>
	<CAPEdYTghm8J20E5COERvG1TidbYNUqcVsjD1TngTyH-zwWE52w@mail.gmail.com>
	<CAC7ttFGRJQAFn0vC3sCC1tguvWmZ==BpKR4i-=UWjDPRgGx-hA@mail.gmail.com>
	<9589f6e5d757412ba5d620cf8e2bf26c@imladris.sermotec.local>
	<56E10D5D.1000303@zg.t-com.hr>
	<38494249596647c6a08591eca41e3000@imladris.sermotec.local>
	<CAC7ttFFbv9YAxXNg7zjc1B9cxLS74v6f6HfBpSt09sULor6EoA@mail.gmail.com>
	<CAHtxdDdY2JRqVuw-W0b+UF0Z_ZUGbCj_iUFUOXi=nrxDDzKb9A@mail.gmail.com>
	<CAC7ttFErJO-gTLBQTHy_6zCcRWBxOYgsXtJXqMC0U8Tv+79AOg@mail.gmail.com>
	<3ED4D03E-7C8C-4DE6-9DC1-1C6E039C5355@jerris.com>
	<CAC7ttFFfGFD+mzKRZWvwNy3BYoKK7FXY1pBgPGUMbj79qYeMmA@mail.gmail.com>
	<CAEwTsAgU8dn+BqO0K94PPRvqTOKDZFBHdgw51tmkfKSrtmzbXA@mail.gmail.com>
	<CAFmZFiPE_9gHbwquw=9-k2WbJMs1CBDV3hmRodX60dKmWJFUbA@mail.gmail.com>
Message-ID: <CAC7ttFFBjjPVeyHc_8Y06dJPbKcxWQzCBBKWJgQL8WezBCxbZw@mail.gmail.com>

Here is more up to date version. It is about 3 weeks old. It si compiled
without sounds.

https://1drv.ms/u/s!Ar-XJmRaByE_rQYqTy7EOFRIuhZM



2016-06-03 0:54 GMT+02:00 Abaci B <abaci64 at gmail.com>:

> any updates on getting updated windows binaries, or is there another place
> to download a more up to date version?
>
>
> On Wed, Apr 6, 2016 at 12:01 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> I'll work on getting new ones out there, but it's easy to build it
>> yourself
>>
>>
>> On Wednesday, April 6, 2016, Gregor Nanger <gregor at infomedia.si> wrote:
>>
>>> Thank you Michael!
>>>
>>> Maybe it is something with my configuration of VS...
>>>
>>> Can you copy your build also on
>>> http://files.freeswitch.org/windows/installer/x64/?
>>>
>>> 2016-04-06 1:08 GMT+02:00 Michael Jerris <mike at jerris.com>:
>>>
>>>> I just built windows fine yesterday.  It may be unhappy with rebuild,
>>>> but I built fine. (this should cover master).... mod_avmd might have been
>>>> failing for a few commits, but I pushed the fix for that module yesterday,
>>>> all the others in default configuration built fine.
>>>>
>>>> On Apr 5, 2016, at 6:38 PM, Gregor Nanger <gregor at infomedia.si> wrote:
>>>>
>>>> Just bumping this post if someone has any hints.
>>>>
>>>> I am trying to build Freeswitch with Visual studio 2015, but keep
>>>> getting error:
>>>> Error MSB4057 The target "v8:Rebuild" does not exist in the project.
>>>> [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln] libv8
>>>> ? ?
>>>> C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
>>>> ?"
>>>>
>>>>
>>>> Would realy like to successfuly build Freeswitch in windows. I tried
>>>> with 1.6 branch.
>>>>
>>>> Best regards, Gregor
>>>>
>>>> 2016-03-17 5:43 GMT+01:00 Sergey Safarov <s.safarov at gmail.com>:
>>>>
>>>>> One week ago I successfully compiled mod_V8 on CentOS 7.
>>>>> May be switch to Linux?
>>>>>
>>>>> On Thu, Mar 17, 2016 at 12:49 AM, Gregor Nanger <gregor at infomedia.si>
>>>>> wrote:
>>>>>
>>>>>> Thank you Harald.
>>>>>>
>>>>>> I tried with latest branch 1.7 and x64 and got errors regarding
>>>>>> mod_V8. I tried what Peter suggested, but I am more in c# and web projects
>>>>>> and do not have experience in building  C++ projects. There is already
>>>>>> prebuilt setup on freeswitch site, so someone successfully build it :-)) It
>>>>>> is not so important for us at this point to make own build, so will try
>>>>>> again later.
>>>>>>
>>>>>>
>>>>>> 2016-03-11 7:32 GMT+01:00 Harald Petrovitsch <
>>>>>> Harald.Petrovitsch at sermotec.at>:
>>>>>>
>>>>>>> Hi Gregor,
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> I only do a
>>>>>>>
>>>>>>> Git.exe clone ?bv1.6
>>>>>>> https://freeswitch.org/stash/scm/fs/freeswitch.git .
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Loaded the solution into vs2015, set configuration Win32 / Release
>>>>>>> and press f7 (need to do it two times)
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> The build ended with
>>>>>>>
>>>>>>> ========== Build: 20 succeeded, 0 failed, 157 up-to-date, 15 skipped
>>>>>>> ==========
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> I?ve attached a list of the generated mod folder
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Regards
>>>>>>>
>>>>>>> Harald
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>>>>>> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *
>>>>>>> Shishko
>>>>>>> *Gesendet:* Donnerstag, 10. M?rz 2016 07:00
>>>>>>> *An:* freeswitch-users at lists.freeswitch.org
>>>>>>>
>>>>>>> *Betreff:* Re: [Freeswitch-users] Windows build
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Hi Harald,
>>>>>>>
>>>>>>> what did you do to build libv8 and mod_v8? I tried with VS2015
>>>>>>> Update 1, branch 1.6, but to no avail.
>>>>>>>
>>>>>>> Thanks
>>>>>>>
>>>>>>> On 03/07/2016 08:20 AM, Harald Petrovitsch wrote:
>>>>>>>
>>>>>>> Hi Gregor !
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> V8 libs and mod builds fine here (visual Studio 2015 Sp1, 1.6
>>>>>>> branch, used tortoiseGit to download it))
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Regards
>>>>>>>
>>>>>>> Harald
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> *Von:* freeswitch-users-bounces at lists.freeswitch.org [
>>>>>>> mailto:freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag v**on
>>>>>>> *Gregor Nanger
>>>>>>> *Gesendet:* Montag, 07. M?rz 2016 00:19
>>>>>>> *An:* FreeSWITCH Users Help
>>>>>>> *Betreff:* Re: [Freeswitch-users] Windows build
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Thank you, H
>>>>>>>
>>>>>>> ?a?
>>>>>>>
>>>>>>> rald. This works: "AFAIK, for the ?'lame/lame.h'? you have to change
>>>>>>> the include line to  only ?lame.h?"
>>>>>>>
>>>>>>> B
>>>>>>>
>>>>>>> ?ut for v8 stil do  not have solution. I do not want to exclude
>>>>>>> mod_v8, since this module runs javascript. But, can you please confirm me
>>>>>>> that is not yet compatible, to stop trying to solve it.
>>>>>>>
>>>>>>>
>>>>>>> Any other suggestion what does this mean:
>>>>>>> "
>>>>>>> Error MSB4057 The target "v8:Rebuild" does not exist in the project.
>>>>>>> [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln] libv8
>>>>>>>
>>>>>>> ? ?
>>>>>>>
>>>>>>> C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
>>>>>>>
>>>>>>> ?"
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Best regards, Gregor?
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> 2016-03-06 16:39 GMT+01:00 Peter Olsson <peter at olssononline.se>:
>>>>>>>
>>>>>>> Remove mod_v8 from the build. I don't think it's compatible with
>>>>>>> VS2015 for now. However, all other modules should be ok.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> /Peter
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> 2016-03-06 12:31 GMT+01:00 Gregor Nanger <gregor at infomedia.si>:
>>>>>>>
>>>>>>> ?This helped a lot, thank you. Now I have only few errors. Any hint?
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Error MSB4057 The target "v8:Rebuild" does not exist in the project.
>>>>>>> [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln]
>>>>>>> libv8 C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
>>>>>>>
>>>>>>> Error MSB4057 The target "v8:Rebuild" does not exist in the project.
>>>>>>> [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln] libv8
>>>>>>> C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
>>>>>>>
>>>>>>> Error C1083 Cannot open include file: 'lame/lame.h': No such file or
>>>>>>> directory mod_shout C:\Git\freeswitch\src\mod\formats\mod_shout\mod_shout.c
>>>>>>> 38
>>>>>>>
>>>>>>> Error LNK1181 cannot open input file 'icui18n.lib' mod_v8
>>>>>>> C:\Git\freeswitch\src\mod\languages\mod_v8\LINK 1
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> ?
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> 2016-03-06 7:53 GMT+01:00 Peter Olsson <peter at olssononline.se>:
>>>>>>>
>>>>>>> One common mistake is that you allow Git to modify line endings.
>>>>>>> Make sure autocrlf is turned off - then clone the repository again from
>>>>>>> scratch.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Also, I'm not sure if it will work in VS2015, but give it a try.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> /Peter
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> 2016-03-06 2:06 GMT+01:00 Gregor Nanger <gregor at infomedia.si>:
>>>>>>>
>>>>>>> Hi!
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> I want to build Freeswitch on windows with visual studio 2015.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Where should I start if I get 600 errors when try to Rebuild All. I
>>>>>>> opened solution and start Rebuild All, but I get so many errors that I
>>>>>>> belive that I am doing something wrong.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Mainl yre errors regarding:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Cannot open source file....
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Best regards, Gregor
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>>
>>>>>>> *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
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>>
>>>>>>> *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
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>>
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>
>>>>>>> consulting at freeswitch.org
>>>>>>>
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>>
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>> http://confluence.freeswitch.org
>>>>>>>
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> 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
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> 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
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> 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
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
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
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From krice at freeswitch.org  Fri Jun  3 04:21:42 2016
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 2 Jun 2016 19:21:42 -0500
Subject: [Freeswitch-users] Windows build
In-Reply-To: <CAC7ttFFBjjPVeyHc_8Y06dJPbKcxWQzCBBKWJgQL8WezBCxbZw@mail.gmail.com>
References: <CAC7ttFFYf+dt3NfgRQy0R5PbNut0=-=mmxv6+gTvUYFB1RsKBA@mail.gmail.com>
	<CAPEdYThObr1ha7ffrFNijAj3H30Z6Zk41TETEgdoeZWxLu6igg@mail.gmail.com>
	<CAC7ttFE98byMZX+uNYUcLJ-A8+WnQwcwXBqqPO-g2fBu3XRfvA@mail.gmail.com>
	<CAPEdYTghm8J20E5COERvG1TidbYNUqcVsjD1TngTyH-zwWE52w@mail.gmail.com>
	<CAC7ttFGRJQAFn0vC3sCC1tguvWmZ==BpKR4i-=UWjDPRgGx-hA@mail.gmail.com>
	<9589f6e5d757412ba5d620cf8e2bf26c@imladris.sermotec.local>
	<56E10D5D.1000303@zg.t-com.hr>
	<38494249596647c6a08591eca41e3000@imladris.sermotec.local>
	<CAC7ttFFbv9YAxXNg7zjc1B9cxLS74v6f6HfBpSt09sULor6EoA@mail.gmail.com>
	<CAHtxdDdY2JRqVuw-W0b+UF0Z_ZUGbCj_iUFUOXi=nrxDDzKb9A@mail.gmail.com>
	<CAC7ttFErJO-gTLBQTHy_6zCcRWBxOYgsXtJXqMC0U8Tv+79AOg@mail.gmail.com>
	<3ED4D03E-7C8C-4DE6-9DC1-1C6E039C5355@jerris.com>
	<CAC7ttFFfGFD+mzKRZWvwNy3BYoKK7FXY1pBgPGUMbj79qYeMmA@mail.gmail.com>
	<CAEwTsAgU8dn+BqO0K94PPRvqTOKDZFBHdgw51tmkfKSrtmzbXA@mail.gmail.com>
	<CAFmZFiPE_9gHbwquw=9-k2WbJMs1CBDV3hmRodX60dKmWJFUbA@mail.gmail.com>
	<CAC7ttFFBjjPVeyHc_8Y06dJPbKcxWQzCBBKWJgQL8WezBCxbZw@mail.gmail.com>
Message-ID: <487E0FE8-B9DB-4760-B0B7-6ECC2219C579@freeswitch.org>

Please do not post links to 3rd party binaries on the lat. this is a major security issue as we can not guarantee the source that these were built with.

Sent from my iPhone

> On Jun 2, 2016, at 6:25 PM, Gregor Nanger <gregor at infomedia.si> wrote:
> 
> Here is more up to date version. It is about 3 weeks old. It si compiled without sounds.
> 
> https://censored
> 
> 
> 
> 2016-06-03 0:54 GMT+02:00 Abaci B <abaci64 at gmail.com>:
>> any updates on getting updated windows binaries, or is there another place to download a more up to date version?
>> 
>> 
>>> On Wed, Apr 6, 2016 at 12:01 PM, Michael Jerris <mike at jerris.com> wrote:
>>> I'll work on getting new ones out there, but it's easy to build it yourself
>>> 
>>> 
>>>> On Wednesday, April 6, 2016, Gregor Nanger <gregor at infomedia.si> wrote:
>>>> Thank you Michael!
>>>> 
>>>> Maybe it is something with my configuration of VS...
>>>> 
>>>> Can you copy your build also on http://files.freeswitch.org/windows/installer/x64/?
>>>> 
>>>> 2016-04-06 1:08 GMT+02:00 Michael Jerris <mike at jerris.com>:
>>>>> I just built windows fine yesterday.  It may be unhappy with rebuild, but I built fine. (this should cover master).... mod_avmd might have been failing for a few commits, but I pushed the fix for that module yesterday, all the others in default configuration built fine.
>>>>> 
>>>>>> On Apr 5, 2016, at 6:38 PM, Gregor Nanger <gregor at infomedia.si> wrote:
>>>>>> 
>>>>>> Just bumping this post if someone has any hints.
>>>>>> 
>>>>>> I am trying to build Freeswitch with Visual studio 2015, but keep getting error:
>>>>>> Error MSB4057 The target "v8:Rebuild" does not exist in the project. [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln] libv8? ?C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1?"
>>>>>> 
>>>>>> Would realy like to successfuly build Freeswitch in windows. I tried with 1.6 branch.
>>>>>> 
>>>>>> Best regards, Gregor
>>>>>> 
>>>>>> 2016-03-17 5:43 GMT+01:00 Sergey Safarov <s.safarov at gmail.com>:
>>>>>>> One week ago I successfully compiled mod_V8 on CentOS 7.
>>>>>>> May be switch to Linux?
>>>>>>> 
>>>>>>>> On Thu, Mar 17, 2016 at 12:49 AM, Gregor Nanger <gregor at infomedia.si> wrote:
>>>>>>>> Thank you Harald.
>>>>>>>> 
>>>>>>>> I tried with latest branch 1.7 and x64 and got errors regarding mod_V8. I tried what Peter suggested, but I am more in c# and web projects and do not have experience in building  C++ projects. There is already prebuilt setup on freeswitch site, so someone successfully build it :-)) It is not so important for us at this point to make own build, so will try again later.
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 2016-03-11 7:32 GMT+01:00 Harald Petrovitsch <Harald.Petrovitsch at sermotec.at>:
>>>>>>>>> Hi Gregor,
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> I only do a
>>>>>>>>> 
>>>>>>>>> Git.exe clone ?bv1.6 https://freeswitch.org/stash/scm/fs/freeswitch.git .
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> Loaded the solution into vs2015, set configuration Win32 / Release and press f7 (need to do it two times)
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> The build ended with
>>>>>>>>> 
>>>>>>>>> ========== Build: 20 succeeded, 0 failed, 157 up-to-date, 15 skipped ==========
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> I?ve attached a list of the generated mod folder
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> Regards
>>>>>>>>> 
>>>>>>>>> Harald
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Shishko
>>>>>>>>> Gesendet: Donnerstag, 10. M?rz 2016 07:00
>>>>>>>>> An: freeswitch-users at lists.freeswitch.org
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> Betreff: Re: [Freeswitch-users] Windows build
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> Hi Harald,
>>>>>>>>> 
>>>>>>>>> what did you do to build libv8 and mod_v8? I tried with VS2015 Update 1, branch 1.6, but to no avail.
>>>>>>>>> 
>>>>>>>>> Thanks
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> On 03/07/2016 08:20 AM, Harald Petrovitsch wrote:
>>>>>>>>> 
>>>>>>>>> Hi Gregor !
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> V8 libs and mod builds fine here (visual Studio 2015 Sp1, 1.6 branch, used tortoiseGit to download it))
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> Regards
>>>>>>>>> 
>>>>>>>>> Harald
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Gregor Nanger
>>>>>>>>> Gesendet: Montag, 07. M?rz 2016 00:19
>>>>>>>>> An: FreeSWITCH Users Help
>>>>>>>>> Betreff: Re: [Freeswitch-users] Windows build
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> Thank you, H
>>>>>>>>> 
>>>>>>>>> ?a?
>>>>>>>>> 
>>>>>>>>> rald. This works: "AFAIK, for the ?'lame/lame.h'? you have to change the include line to  only ?lame.h?"
>>>>>>>>> 
>>>>>>>>> B
>>>>>>>>> 
>>>>>>>>> ?ut for v8 stil do  not have solution. I do not want to exclude mod_v8, since this module runs javascript. But, can you please confirm me that is not yet compatible, to stop trying to solve it.
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> Any other suggestion what does this mean:
>>>>>>>>> "
>>>>>>>>> Error MSB4057 The target "v8:Rebuild" does not exist in the project. [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln] libv8
>>>>>>>>> 
>>>>>>>>> ? ?
>>>>>>>>> 
>>>>>>>>> C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
>>>>>>>>> 
>>>>>>>>> ?"
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> Best regards, Gregor?
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> 2016-03-06 16:39 GMT+01:00 Peter Olsson <peter at olssononline.se>:
>>>>>>>>> 
>>>>>>>>> Remove mod_v8 from the build. I don't think it's compatible with VS2015 for now. However, all other modules should be ok.
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> /Peter
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> 2016-03-06 12:31 GMT+01:00 Gregor Nanger <gregor at infomedia.si>:
>>>>>>>>> 
>>>>>>>>> ?This helped a lot, thank you. Now I have only few errors. Any hint?
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> Error MSB4057 The target "v8:Rebuild" does not exist in the project. [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln] libv8 C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
>>>>>>>>> 
>>>>>>>>> Error MSB4057 The target "v8:Rebuild" does not exist in the project. [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln] libv8 C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
>>>>>>>>> 
>>>>>>>>> Error C1083 Cannot open include file: 'lame/lame.h': No such file or directory mod_shout C:\Git\freeswitch\src\mod\formats\mod_shout\mod_shout.c 38
>>>>>>>>> 
>>>>>>>>> Error LNK1181 cannot open input file 'icui18n.lib' mod_v8 C:\Git\freeswitch\src\mod\languages\mod_v8\LINK 1
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> ?
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> 2016-03-06 7:53 GMT+01:00 Peter Olsson <peter at olssononline.se>:
>>>>>>>>> 
>>>>>>>>> One common mistake is that you allow Git to modify line endings. Make sure autocrlf is turned off - then clone the repository again from scratch.
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> Also, I'm not sure if it will work in VS2015, but give it a try.
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> /Peter
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> 2016-03-06 2:06 GMT+01:00 Gregor Nanger <gregor at infomedia.si>:
>>>>>>>>> 
>>>>>>>>> Hi!
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> I want to build Freeswitch on windows with visual studio 2015. 
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> Where should I start if I get 600 errors when try to Rebuild All. I opened solution and start Rebuild All, but I get so many errors that I belive that I am doing something wrong.
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> Mainl yre errors regarding:
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> Cannot open source file....
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> Best regards, Gregor
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>> 
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>> http://www.cluecon.com
>>>>>>>>> 
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>> 
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>> http://www.cluecon.com
>>>>>>>>> 
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> --
>>>>>>>>> 
>>>>>>>>> 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
>>>>>>>>> 
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>> 
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>> http://www.cluecon.com
>>>>>>>>> 
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>> 
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>> http://www.cluecon.com
>>>>>>>>> 
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> --
>>>>>>>>> 
>>>>>>>>> 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
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services: 
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>  
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>> http://www.cluecon.com
>>>>>>>>>  
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>  
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>> 
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>> http://www.cluecon.com
>>>>>>>>> 
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> -- 
>>>>>>>> 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
>>>>>>>> 
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>> 
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>> 
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> 
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>> 
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> -- 
>>>>>> 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
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services: 
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> 
>>>> -- 
>>>> 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
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> -- 
> 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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From shishko69 at gmail.com  Fri Jun  3 10:23:06 2016
From: shishko69 at gmail.com (Denis Papes)
Date: Fri, 03 Jun 2016 06:23:06 +0000
Subject: [Freeswitch-users] Lua Script Does Not Get Executed
In-Reply-To: <CAPrxTnpVFh8fzaqPzMKykdq5f7s2Pvktg2amNMSzS3tyw7JChg@mail.gmail.com>
References: <CAPrxTnpVFh8fzaqPzMKykdq5f7s2Pvktg2amNMSzS3tyw7JChg@mail.gmail.com>
Message-ID: <CAHT_yt8iUuwJy71cTjqey=gszQDTq805dUYC0HRAkN=SViTr0w@mail.gmail.com>

First check if user under which FreeSWITCH is running has permissions to
access /home/danieln/, then check if /home/danieln/checkVoicemails2 has
exec permission (if not, chmod +x /home/danieln/checkVoicemails2).


On Thu, Jun 2, 2016 at 11:37 PM Cody Rosenbloom <cody.rosenbloom at gmail.com>
wrote:

> Hi,
>
> I am very new to freeswitch and Lua. I'm trying to get a lua script which
> I've set up to send voicemails but as far as I can tell, the script is
> never even executed. This is the basic sequence of events.
>
> -> Caller leaves a voicemail. This is stored on the server, no problem.
> ->After voicemail is stored, I call a lua script passing the email id of
> the user and his extension
> ->The os.execute command within the script passes these values to a simple
> shell script which grabs the voicemail audio file, logs some debug output
> to an output file so I can verify its been run and then sends me an email.
> When I test the script from fs_cli console using
>
> lua <scriptName> <args> , it works fine and I see the debug output logged.
> When I actually call and leave a voicemail, this does not happen. Can
> someone point out what I'm doing wrong ?
>
> My xml and lua script are pasted at http://pastebin.com/2N6LcfuV
>
> Thank you very much!
>
> Cody
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From gregor at infomedia.si  Fri Jun  3 10:32:51 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Fri, 03 Jun 2016 06:32:51 +0000
Subject: [Freeswitch-users] Windows build
In-Reply-To: <487E0FE8-B9DB-4760-B0B7-6ECC2219C579@freeswitch.org>
References: <CAC7ttFFYf+dt3NfgRQy0R5PbNut0=-=mmxv6+gTvUYFB1RsKBA@mail.gmail.com>
	<CAPEdYThObr1ha7ffrFNijAj3H30Z6Zk41TETEgdoeZWxLu6igg@mail.gmail.com>
	<CAC7ttFE98byMZX+uNYUcLJ-A8+WnQwcwXBqqPO-g2fBu3XRfvA@mail.gmail.com>
	<CAPEdYTghm8J20E5COERvG1TidbYNUqcVsjD1TngTyH-zwWE52w@mail.gmail.com>
	<CAC7ttFGRJQAFn0vC3sCC1tguvWmZ==BpKR4i-=UWjDPRgGx-hA@mail.gmail.com>
	<9589f6e5d757412ba5d620cf8e2bf26c@imladris.sermotec.local>
	<56E10D5D.1000303@zg.t-com.hr>
	<38494249596647c6a08591eca41e3000@imladris.sermotec.local>
	<CAC7ttFFbv9YAxXNg7zjc1B9cxLS74v6f6HfBpSt09sULor6EoA@mail.gmail.com>
	<CAHtxdDdY2JRqVuw-W0b+UF0Z_ZUGbCj_iUFUOXi=nrxDDzKb9A@mail.gmail.com>
	<CAC7ttFErJO-gTLBQTHy_6zCcRWBxOYgsXtJXqMC0U8Tv+79AOg@mail.gmail.com>
	<3ED4D03E-7C8C-4DE6-9DC1-1C6E039C5355@jerris.com>
	<CAC7ttFFfGFD+mzKRZWvwNy3BYoKK7FXY1pBgPGUMbj79qYeMmA@mail.gmail.com>
	<CAEwTsAgU8dn+BqO0K94PPRvqTOKDZFBHdgw51tmkfKSrtmzbXA@mail.gmail.com>
	<CAFmZFiPE_9gHbwquw=9-k2WbJMs1CBDV3hmRodX60dKmWJFUbA@mail.gmail.com>
	<CAC7ttFFBjjPVeyHc_8Y06dJPbKcxWQzCBBKWJgQL8WezBCxbZw@mail.gmail.com>
	<487E0FE8-B9DB-4760-B0B7-6ECC2219C579@freeswitch.org>
Message-ID: <CAC7ttFEz8TX_mCBT_yVry_n5dy53YkswZgzUHBJzpomCu9SVUg@mail.gmail.com>

Oh, I understand. Just wanted to help. Guess you cannot delete post, so
Google won't index it?

Anyway, it is not virus.

On Fri, Jun 3, 2016, 02:23 Ken Rice <krice at freeswitch.org> wrote:

> Please do not post links to 3rd party binaries on the lat. this is a major
> security issue as we can not guarantee the source that these were built
> with.
>
> Sent from my iPhone
>
> On Jun 2, 2016, at 6:25 PM, Gregor Nanger <gregor at infomedia.si> wrote:
>
> Here is more up to date version. It is about 3 weeks old. It si compiled
> without sounds.
>
> https:// <https://1drv.ms/u/s!Ar-XJmRaByE_rQYqTy7EOFRIuhZM>censored
>
>
>
> 2016-06-03 0:54 GMT+02:00 Abaci B <abaci64 at gmail.com>:
>
>> any updates on getting updated windows binaries, or is there another
>> place to download a more up to date version?
>>
>>
>> On Wed, Apr 6, 2016 at 12:01 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> I'll work on getting new ones out there, but it's easy to build it
>>> yourself
>>>
>>>
>>> On Wednesday, April 6, 2016, Gregor Nanger <gregor at infomedia.si> wrote:
>>>
>>>> Thank you Michael!
>>>>
>>>> Maybe it is something with my configuration of VS...
>>>>
>>>> Can you copy your build also on
>>>> http://files.freeswitch.org/windows/installer/x64/?
>>>>
>>>> 2016-04-06 1:08 GMT+02:00 Michael Jerris <mike at jerris.com>:
>>>>
>>>>> I just built windows fine yesterday.  It may be unhappy with rebuild,
>>>>> but I built fine. (this should cover master).... mod_avmd might have been
>>>>> failing for a few commits, but I pushed the fix for that module yesterday,
>>>>> all the others in default configuration built fine.
>>>>>
>>>>> On Apr 5, 2016, at 6:38 PM, Gregor Nanger <gregor at infomedia.si> wrote:
>>>>>
>>>>> Just bumping this post if someone has any hints.
>>>>>
>>>>> I am trying to build Freeswitch with Visual studio 2015, but keep
>>>>> getting error:
>>>>> Error MSB4057 The target "v8:Rebuild" does not exist in the project.
>>>>> [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln] libv8
>>>>> ? ?
>>>>> C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
>>>>> ?"
>>>>>
>>>>>
>>>>> Would realy like to successfuly build Freeswitch in windows. I tried
>>>>> with 1.6 branch.
>>>>>
>>>>> Best regards, Gregor
>>>>>
>>>>> 2016-03-17 5:43 GMT+01:00 Sergey Safarov <s.safarov at gmail.com>:
>>>>>
>>>>>> One week ago I successfully compiled mod_V8 on CentOS 7.
>>>>>> May be switch to Linux?
>>>>>>
>>>>>> On Thu, Mar 17, 2016 at 12:49 AM, Gregor Nanger <gregor at infomedia.si>
>>>>>> wrote:
>>>>>>
>>>>>>> Thank you Harald.
>>>>>>>
>>>>>>> I tried with latest branch 1.7 and x64 and got errors regarding
>>>>>>> mod_V8. I tried what Peter suggested, but I am more in c# and web projects
>>>>>>> and do not have experience in building  C++ projects. There is already
>>>>>>> prebuilt setup on freeswitch site, so someone successfully build it :-)) It
>>>>>>> is not so important for us at this point to make own build, so will try
>>>>>>> again later.
>>>>>>>
>>>>>>>
>>>>>>> 2016-03-11 7:32 GMT+01:00 Harald Petrovitsch <
>>>>>>> Harald.Petrovitsch at sermotec.at>:
>>>>>>>
>>>>>>>> Hi Gregor,
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> I only do a
>>>>>>>>
>>>>>>>> Git.exe clone ?bv1.6
>>>>>>>> https://freeswitch.org/stash/scm/fs/freeswitch.git .
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Loaded the solution into vs2015, set configuration Win32 / Release
>>>>>>>> and press f7 (need to do it two times)
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> The build ended with
>>>>>>>>
>>>>>>>> ========== Build: 20 succeeded, 0 failed, 157 up-to-date, 15
>>>>>>>> skipped ==========
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> I?ve attached a list of the generated mod folder
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Regards
>>>>>>>>
>>>>>>>> Harald
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>>>>>>> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *
>>>>>>>> Shishko
>>>>>>>> *Gesendet:* Donnerstag, 10. M?rz 2016 07:00
>>>>>>>> *An:* freeswitch-users at lists.freeswitch.org
>>>>>>>>
>>>>>>>> *Betreff:* Re: [Freeswitch-users] Windows build
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Hi Harald,
>>>>>>>>
>>>>>>>> what did you do to build libv8 and mod_v8? I tried with VS2015
>>>>>>>> Update 1, branch 1.6, but to no avail.
>>>>>>>>
>>>>>>>> Thanks
>>>>>>>>
>>>>>>>> On 03/07/2016 08:20 AM, Harald Petrovitsch wrote:
>>>>>>>>
>>>>>>>> Hi Gregor !
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> V8 libs and mod builds fine here (visual Studio 2015 Sp1, 1.6
>>>>>>>> branch, used tortoiseGit to download it))
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Regards
>>>>>>>>
>>>>>>>> Harald
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> *Von:* freeswitch-users-bounces at lists.freeswitch.org [
>>>>>>>> mailto:freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag
>>>>>>>> v**on *Gregor Nanger
>>>>>>>> *Gesendet:* Montag, 07. M?rz 2016 00:19
>>>>>>>> *An:* FreeSWITCH Users Help
>>>>>>>> *Betreff:* Re: [Freeswitch-users] Windows build
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Thank you, H
>>>>>>>>
>>>>>>>> ?a?
>>>>>>>>
>>>>>>>> rald. This works: "AFAIK, for the ?'lame/lame.h'? you have to
>>>>>>>> change the include line to  only ?lame.h?"
>>>>>>>>
>>>>>>>> B
>>>>>>>>
>>>>>>>> ?ut for v8 stil do  not have solution. I do not want to exclude
>>>>>>>> mod_v8, since this module runs javascript. But, can you please confirm me
>>>>>>>> that is not yet compatible, to stop trying to solve it.
>>>>>>>>
>>>>>>>>
>>>>>>>> Any other suggestion what does this mean:
>>>>>>>> "
>>>>>>>> Error MSB4057 The target "v8:Rebuild" does not exist in the
>>>>>>>> project. [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln] libv8
>>>>>>>>
>>>>>>>> ? ?
>>>>>>>>
>>>>>>>> C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
>>>>>>>>
>>>>>>>> ?"
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Best regards, Gregor?
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> 2016-03-06 16:39 GMT+01:00 Peter Olsson <peter at olssononline.se>:
>>>>>>>>
>>>>>>>> Remove mod_v8 from the build. I don't think it's compatible with
>>>>>>>> VS2015 for now. However, all other modules should be ok.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> /Peter
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> 2016-03-06 12:31 GMT+01:00 Gregor Nanger <gregor at infomedia.si>:
>>>>>>>>
>>>>>>>> ?This helped a lot, thank you. Now I have only few errors. Any
>>>>>>>> hint?
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Error MSB4057 The target "v8:Rebuild" does not exist in the
>>>>>>>> project. [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln]
>>>>>>>> libv8 C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
>>>>>>>>
>>>>>>>> Error MSB4057 The target "v8:Rebuild" does not exist in the
>>>>>>>> project. [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln] libv8
>>>>>>>> C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
>>>>>>>>
>>>>>>>> Error C1083 Cannot open include file: 'lame/lame.h': No such file
>>>>>>>> or directory mod_shout
>>>>>>>> C:\Git\freeswitch\src\mod\formats\mod_shout\mod_shout.c 38
>>>>>>>>
>>>>>>>> Error LNK1181 cannot open input file 'icui18n.lib' mod_v8
>>>>>>>> C:\Git\freeswitch\src\mod\languages\mod_v8\LINK 1
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> ?
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> 2016-03-06 7:53 GMT+01:00 Peter Olsson <peter at olssononline.se>:
>>>>>>>>
>>>>>>>> One common mistake is that you allow Git to modify line endings.
>>>>>>>> Make sure autocrlf is turned off - then clone the repository again from
>>>>>>>> scratch.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Also, I'm not sure if it will work in VS2015, but give it a try.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> /Peter
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> 2016-03-06 2:06 GMT+01:00 Gregor Nanger <gregor at infomedia.si>:
>>>>>>>>
>>>>>>>> Hi!
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> I want to build Freeswitch on windows with visual studio 2015.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Where should I start if I get 600 errors when try to Rebuild All. I
>>>>>>>> opened solution and start Rebuild All, but I get so many errors that I
>>>>>>>> belive that I am doing something wrong.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Mainl yre errors regarding:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Cannot open source file....
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Best regards, Gregor
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>>
>>>>>>>> *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
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>>
>>>>>>>> *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
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>>
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>
>>>>>>>> consulting at freeswitch.org
>>>>>>>>
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> 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
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> 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
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> 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
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> 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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
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
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From david.villasmil.work at gmail.com  Fri Jun  3 12:25:18 2016
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 3 Jun 2016 10:25:18 +0200
Subject: [Freeswitch-users] Ringback after pre_answer without bridge?
In-Reply-To: <CA+GmxbbJDyCmiM9LhGEBq1NLVAPwVT4oXCV=_Dpt5FsiOsvkgw@mail.gmail.com>
References: <CA+GmxbbJDyCmiM9LhGEBq1NLVAPwVT4oXCV=_Dpt5FsiOsvkgw@mail.gmail.com>
Message-ID: <CAFGRPVp0Sz2cdMzX6Ju=vA5JDJ16TUAGyCEuirv6Rq3mJ-x=sg@mail.gmail.com>

Have you tried moving the "pre_answer" to before the ringback?
?

On Tue, May 31, 2016 at 1:57 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:

> Hi all!
>
> I'd like to resolve some unclear ringback behavior in early media
> mode. I'm trying to implement ringback while no bridge active.
> A calls FS, FS does something, then searching for B and bridges A to B.
>
> I'd like to start playing custom ringback immediately after call from
> A gets into FS, not when we start bridge to B.
>
> I'm doing something like that:
>
> <action application="set" data="ringback=%(2000, 4000, 440.0, 480.0)"/>
> <action application="set" data="instant_ringback=true"/>
> <action application="set" data="bridge_early_media=true"/>
> <action application="pre_answer"/>
> <action application="sleep" data="10000"/>
> ...
> <action application="bridge" data="some bridge data"/>
>
> There is no ringback to A in such scenario until bridge is called in
> further dialplan processing. I've tried all combinations of
> ignore/bridge early media, but with no luck. Pre_answer can be called
> directly or by FS when call is recorded, with no difference. There is
> a wiki page that recommends usage of ring_ready:
>
>
> https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+ring_ready
>
> ring_ready works, but it can give only Ringing to calling side, so  I
> can't pass custom RBT to calling party. It would be enough for me, but
> once somebody calls pre_answer, no early media is passed until bridge
> attempt.
>
> What can be done to resolve this? Thanks in advance.
>
>
>
>
>
>
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kbdfck at gmail.com  Fri Jun  3 12:43:37 2016
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Fri, 3 Jun 2016 11:43:37 +0300
Subject: [Freeswitch-users] Ringback after pre_answer without bridge?
In-Reply-To: <CAFGRPVp0Sz2cdMzX6Ju=vA5JDJ16TUAGyCEuirv6Rq3mJ-x=sg@mail.gmail.com>
References: <CA+GmxbbJDyCmiM9LhGEBq1NLVAPwVT4oXCV=_Dpt5FsiOsvkgw@mail.gmail.com>
	<CAFGRPVp0Sz2cdMzX6Ju=vA5JDJ16TUAGyCEuirv6Rq3mJ-x=sg@mail.gmail.com>
Message-ID: <CA+GmxbYaryvepfcRgY07+yws_0E_zgrwprvcXLtRLtkXy1i27w@mail.gmail.com>

I've tried all variations of ringback and pre_answer order.
I've found solution that works for me, but it appears to be a bug in
displace_session or uuid_displace implementation.

Method1:
Thing that works:
<action application="pre_answer"/>
<action application="set" data="send_silence_when_idle=1400"/>
<action application="displace_session"
data="tone_stream://%(1000,4000,425);loops=-1 0 mux}"/>
<action application="sleep" data="8000"/>

Thing that doesn't work: the above plays tone_stream during sleep, but stop
fails
<action application="stop_displace_session"
data="tone_stream://%(1000,4000,425);loops=-1}"/>. It doesn't stop displace.

Method 2. Use uuid_displace instead of displace_session:
<action application="set" data="api_result=${uuid_displace ${uuid} start
tone_stream://L=5;%(1000,4000,425)}"/>
<action application="sleep" data="8000"/>
<action application="set" data="api_result=${uuid_displace ${uuid} stop
tone_stream://L=5;%(1000,4000,425);}"/>

It works, but there is a problem: it incorrectly interprets tone_stream
syntax of this: "tone_stream://%(1000,4000,425);loops=-1" and plays some
strange DTMF (possibly it interpretes some symbols as DTMF) after generated
sound.



2016-06-03 11:25 GMT+03:00 David Villasmil <david.villasmil.work at gmail.com>:

> Have you tried moving the "pre_answer" to before the ringback?
> ?
>
> On Tue, May 31, 2016 at 1:57 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>
>> Hi all!
>>
>> I'd like to resolve some unclear ringback behavior in early media
>> mode. I'm trying to implement ringback while no bridge active.
>> A calls FS, FS does something, then searching for B and bridges A to B.
>>
>> I'd like to start playing custom ringback immediately after call from
>> A gets into FS, not when we start bridge to B.
>>
>> I'm doing something like that:
>>
>> <action application="set" data="ringback=%(2000, 4000, 440.0, 480.0)"/>
>> <action application="set" data="instant_ringback=true"/>
>> <action application="set" data="bridge_early_media=true"/>
>> <action application="pre_answer"/>
>> <action application="sleep" data="10000"/>
>> ...
>> <action application="bridge" data="some bridge data"/>
>>
>> There is no ringback to A in such scenario until bridge is called in
>> further dialplan processing. I've tried all combinations of
>> ignore/bridge early media, but with no luck. Pre_answer can be called
>> directly or by FS when call is recorded, with no difference. There is
>> a wiki page that recommends usage of ring_ready:
>>
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+ring_ready
>>
>> ring_ready works, but it can give only Ringing to calling side, so  I
>> can't pass custom RBT to calling party. It would be enough for me, but
>> once somebody calls pre_answer, no early media is passed until bridge
>> attempt.
>>
>> What can be done to resolve this? Thanks in advance.
>>
>>
>>
>>
>>
>>
>>
>> --
>> Best regards,
>>
>> Dmitry Sytchev,
>> IT Engineer
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From carlosj.gf at gmail.com  Fri Jun  3 13:37:22 2016
From: carlosj.gf at gmail.com (=?UTF-8?Q?Carlos_Gonz=C3=A1lez_Florido?=)
Date: Fri, 3 Jun 2016 11:37:22 +0200
Subject: [Freeswitch-users] What mod_verto implementations are out in
	the wild?
In-Reply-To: <CAJ0sghRiRmsCzx0Vm2W3EKG6T_b7a_29YZ=RH+RkmvpSTp5u+w@mail.gmail.com>
References: <CAJ0sghRiRmsCzx0Vm2W3EKG6T_b7a_29YZ=RH+RkmvpSTp5u+w@mail.gmail.com>
Message-ID: <CAFKjMAasGUvwk9iuWsfKMFTFG5gT-V1Nbh34AgSfqUt17K6zEA@mail.gmail.com>

Hi.

We developed a full Verto client and server for the NkMEDIA
<https://github.com/NetComposer/nkmedia/blob/master/README.md> project (in
Erlang).

Carlos

On Thu, Jun 2, 2016 at 8:08 PM, Chad Phillips <chad at apartmentlines.com>
wrote:
>
> Curious if anyone knows of other well-developed mod_verto implementations
besides Verto Communicator. Documentation is still fairly thin, and it's
capable of a lot, so I'm wondering who else has climbed the mountain. :)
>
> I have a fairly robust implementation running on
https://connect.circleanywhere.com for our 'main room' videoconference.
>
> Not sure how useful a list of implementations would be for others,
perhaps we could add it to the mod_verto wiki page?
>
> Chad
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From colton.conor at gmail.com  Fri Jun  3 16:58:44 2016
From: colton.conor at gmail.com (Colton Conor)
Date: Fri, 3 Jun 2016 07:58:44 -0500
Subject: [Freeswitch-users] SwitchPi
In-Reply-To: <04ea01d1bcf1$c1a373f0$44ea5bd0$@freeswitch.org>
References: <CABV9BUu8ygpbbGLzxmzG70NsiS7dj5D4WRkNaQH6Hzex_Q38yg@mail.gmail.com>
	<00bb01d1bc20$ab399cb0$01acd610$@freeswitch.org>
	<CABV9BUuns9Z+-KYJvj8LeRrDUkKBnL-Dh46dXJ-Y2kDFent2Rg@mail.gmail.com>
	<04ea01d1bcf1$c1a373f0$44ea5bd0$@freeswitch.org>
Message-ID: <CAMDdSzMj5-GU1M1930ueDFGMDiZPTwUqgVFr=Gj2ukGxT2-W2w@mail.gmail.com>

May I ask the question of why make something customer just to run on the
Rasberry Pi? There are X86 SOC coming to the market that will be close to
the cost of the Pi, and be much more powerful. Plus, you can run regular
x86 operating systems.

On Thu, Jun 2, 2016 at 12:11 PM, Ken Rice <krice at freeswitch.org> wrote:

> The custom boards allow for hardware DTMF keypad interface, hook switch
> and a few other things like LCD for CLID on old school phones with that
> provision.
>
>
>
> There will be official Raspian packages sooner or later? theres a know
> issue with the cowbuilder that?s blocking this at this time, we?re working
> on a work around?
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *William
> Scott
> *Sent:* Wednesday, June 1, 2016 6:52 PM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] SwitchPi
>
>
>
> Hi
>
>
>
> Yes, I have FusionPBX/freeSwitch running on a RPi3. Will the ARM build
> ever become "Official"?
>
>
>
> I've watched a 'tube video or two. What is the functionality of the custom
> boards?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Alexander.Haugg at c4b.de  Fri Jun  3 17:09:33 2016
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Fri, 3 Jun 2016 13:09:33 +0000
Subject: [Freeswitch-users] Dialplan comand bridge => Keep the first leg
 open if the callee was hang up the call
Message-ID: <B60DA91F5EA0E24A96E73CDE43CC9EC8C0062A41@vc4bex2.corp.c4b.de>

Hi,

Call leg A is bridged to B
<action application="bridge" data="sofia/gateway/mySipTrunk/B" />

B a pickup the call and hang up the call.
A will hang up too, why?
Is there a possibility, to keep  the first leg open?

I tried park_after_bridge=true, hangup_after_bridge=fals, park as dialplan application after the bridge,
But nothing helps.

Thanks a lot
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From nishadmusthafa at gmail.com  Fri Jun  3 17:11:58 2016
From: nishadmusthafa at gmail.com (Nishad Musthafa)
Date: Fri, 03 Jun 2016 13:11:58 +0000
Subject: [Freeswitch-users] Dialplan comand bridge => Keep the first leg
 open if the callee was hang up the call
In-Reply-To: <B60DA91F5EA0E24A96E73CDE43CC9EC8C0062A41@vc4bex2.corp.c4b.de>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC8C0062A41@vc4bex2.corp.c4b.de>
Message-ID: <CADeOUzs7vP1ndgyWr3QZic840LuiN9XZO-nEKu-5D3NOdmGQxQ@mail.gmail.com>

Are there any elements after this one in the dialplan? If not I think the
expected behaviour is for freeswitch to hangup the call.

On Fri, Jun 3, 2016 at 6:39 PM Alexander Haugg <Alexander.Haugg at c4b.de>
wrote:

> Hi,
>
>
>
> Call leg A is bridged to B
>
> <action application="bridge" data="sofia/gateway/mySipTrunk/B" />
>
>
>
> B a pickup the call and hang up the call.
>
> A will hang up too, why?
>
> Is there a possibility, to keep  the first leg open?
>
>
>
> I tried park_after_bridge=true, hangup_after_bridge=fals, park as dialplan
> application after the bridge,
>
> But nothing helps.
>
>
>
> Thanks a lot
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From nishadmusthafa at gmail.com  Fri Jun  3 17:13:51 2016
From: nishadmusthafa at gmail.com (Nishad Musthafa)
Date: Fri, 03 Jun 2016 13:13:51 +0000
Subject: [Freeswitch-users] Dialplan comand bridge => Keep the first leg
 open if the callee was hang up the call
In-Reply-To: <CADeOUzs7vP1ndgyWr3QZic840LuiN9XZO-nEKu-5D3NOdmGQxQ@mail.gmail.com>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC8C0062A41@vc4bex2.corp.c4b.de>
	<CADeOUzs7vP1ndgyWr3QZic840LuiN9XZO-nEKu-5D3NOdmGQxQ@mail.gmail.com>
Message-ID: <CADeOUzue1qEt8pLQUAncG1NOy2tdD6wemhZWPc8kTtCFx=BSGQ@mail.gmail.com>

I meant expected behaviour with hangup_after_bridge set to false
On Fri, Jun 3, 2016 at 6:41 PM Nishad Musthafa <nishadmusthafa at gmail.com>
wrote:

> Are there any elements after this one in the dialplan? If not I think the
> expected behaviour is for freeswitch to hangup the call.
>
> On Fri, Jun 3, 2016 at 6:39 PM Alexander Haugg <Alexander.Haugg at c4b.de>
> wrote:
>
>> Hi,
>>
>>
>>
>> Call leg A is bridged to B
>>
>> <action application="bridge" data="sofia/gateway/mySipTrunk/B" />
>>
>>
>>
>> B a pickup the call and hang up the call.
>>
>> A will hang up too, why?
>>
>> Is there a possibility, to keep  the first leg open?
>>
>>
>>
>> I tried park_after_bridge=true, hangup_after_bridge=fals, park as
>> dialplan application after the bridge,
>>
>> But nothing helps.
>>
>>
>>
>> Thanks a lot
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
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From Alexander.Haugg at c4b.de  Fri Jun  3 17:20:22 2016
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Fri, 3 Jun 2016 13:20:22 +0000
Subject: [Freeswitch-users] Dialplan comand bridge => Keep the first leg
 open if the callee was hang up the call
In-Reply-To: <CADeOUzue1qEt8pLQUAncG1NOy2tdD6wemhZWPc8kTtCFx=BSGQ@mail.gmail.com>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC8C0062A41@vc4bex2.corp.c4b.de>
	<CADeOUzs7vP1ndgyWr3QZic840LuiN9XZO-nEKu-5D3NOdmGQxQ@mail.gmail.com>
	<CADeOUzue1qEt8pLQUAncG1NOy2tdD6wemhZWPc8kTtCFx=BSGQ@mail.gmail.com>
Message-ID: <B60DA91F5EA0E24A96E73CDE43CC9EC8C0062A64@vc4bex2.corp.c4b.de>

See again:
I tried park_after_bridge=true, hangup_after_bridge=fals, park as dialplan application after the bridge,


Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Nishad Musthafa
Gesendet: Freitag, 3. Juni 2016 15:14
An: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Betreff: Re: [Freeswitch-users] Dialplan comand bridge => Keep the first leg open if the callee was hang up the call

I meant expected behaviour with hangup_after_bridge set to false
On Fri, Jun 3, 2016 at 6:41 PM Nishad Musthafa <nishadmusthafa at gmail.com<mailto:nishadmusthafa at gmail.com>> wrote:
Are there any elements after this one in the dialplan? If not I think the expected behaviour is for freeswitch to hangup the call.

On Fri, Jun 3, 2016 at 6:39 PM Alexander Haugg <Alexander.Haugg at c4b.de<mailto:Alexander.Haugg at c4b.de>> wrote:
Hi,

Call leg A is bridged to B
<action application="bridge" data="sofia/gateway/mySipTrunk/B" />

B a pickup the call and hang up the call.
A will hang up too, why?
Is there a possibility, to keep  the first leg open?

I tried park_after_bridge=true, hangup_after_bridge=fals, park as dialplan application after the bridge,
But nothing helps.

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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From Alexander.Haugg at c4b.de  Fri Jun  3 17:52:12 2016
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Fri, 3 Jun 2016 13:52:12 +0000
Subject: [Freeswitch-users] Dialplan comand bridge => Keep the first leg
 open if the callee was hang up the call
In-Reply-To: <B60DA91F5EA0E24A96E73CDE43CC9EC8C0062A64@vc4bex2.corp.c4b.de>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC8C0062A41@vc4bex2.corp.c4b.de>
	<CADeOUzs7vP1ndgyWr3QZic840LuiN9XZO-nEKu-5D3NOdmGQxQ@mail.gmail.com>
	<CADeOUzue1qEt8pLQUAncG1NOy2tdD6wemhZWPc8kTtCFx=BSGQ@mail.gmail.com>
	<B60DA91F5EA0E24A96E73CDE43CC9EC8C0062A64@vc4bex2.corp.c4b.de>
Message-ID: <B60DA91F5EA0E24A96E73CDE43CC9EC8C0062A77@vc4bex2.corp.c4b.de>

Ok, after a tiny look into the switch_ivr_bridge?
The solution is the channel variable hold_hangup_xfer_exten

Thanks

Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Alexander Haugg
Gesendet: Freitag, 3. Juni 2016 15:20
An: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Betreff: Re: [Freeswitch-users] Dialplan comand bridge => Keep the first leg open if the callee was hang up the call

See again:
I tried park_after_bridge=true, hangup_after_bridge=fals, park as dialplan application after the bridge,


Von: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Nishad Musthafa
Gesendet: Freitag, 3. Juni 2016 15:14
An: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Betreff: Re: [Freeswitch-users] Dialplan comand bridge => Keep the first leg open if the callee was hang up the call

I meant expected behaviour with hangup_after_bridge set to false
On Fri, Jun 3, 2016 at 6:41 PM Nishad Musthafa <nishadmusthafa at gmail.com<mailto:nishadmusthafa at gmail.com>> wrote:
Are there any elements after this one in the dialplan? If not I think the expected behaviour is for freeswitch to hangup the call.

On Fri, Jun 3, 2016 at 6:39 PM Alexander Haugg <Alexander.Haugg at c4b.de<mailto:Alexander.Haugg at c4b.de>> wrote:
Hi,

Call leg A is bridged to B
<action application="bridge" data="sofia/gateway/mySipTrunk/B" />

B a pickup the call and hang up the call.
A will hang up too, why?
Is there a possibility, to keep  the first leg open?

I tried park_after_bridge=true, hangup_after_bridge=fals, park as dialplan application after the bridge,
But nothing helps.

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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From krice at freeswitch.org  Fri Jun  3 18:13:39 2016
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 3 Jun 2016 09:13:39 -0500
Subject: [Freeswitch-users] SwitchPi
In-Reply-To: <CAMDdSzMj5-GU1M1930ueDFGMDiZPTwUqgVFr=Gj2ukGxT2-W2w@mail.gmail.com>
References: <CABV9BUu8ygpbbGLzxmzG70NsiS7dj5D4WRkNaQH6Hzex_Q38yg@mail.gmail.com>	<00bb01d1bc20$ab399cb0$01acd610$@freeswitch.org>	<CABV9BUuns9Z+-KYJvj8LeRrDUkKBnL-Dh46dXJ-Y2kDFent2Rg@mail.gmail.com>	<04ea01d1bcf1$c1a373f0$44ea5bd0$@freeswitch.org>
	<CAMDdSzMj5-GU1M1930ueDFGMDiZPTwUqgVFr=Gj2ukGxT2-W2w@mail.gmail.com>
Message-ID: <095c01d1bda2$2159f430$640ddc90$@freeswitch.org>

Why the Pi you ask? Why custom hardware to go with it? 

 

Generic X86 Hardware is not the end all be all. While the X86 based CPUs means that there can be less portability issues than say something like ARM or SPARC or other architectures, the PI still typically offers several other capabilities that may not be offered on the other Single Board Computers at a $35 price point.

 

Then there is the well documented pile of GPIO ports that make it quite easy to drive things like custom keypads, servos, steppers, etc etc etc?  this means you can spin a small custom board (in my case its about 1.5 square inches) that has the specific additional hardware for adding an additional 24 GPIOs giving room for a combination of something like 36 buttons worth of keypads and 12 other dry contact inputs and outputs (if you use 12 of the GPIO pins to drive a keypad array you can get 36 buttons. Basic formula  for max number of potential keys   (n/2)^2 where n is the number of GPIO pins used in the keypad array. 

 

Now mix that with that fact that ARM is not a second class Linux citizen anymore, we can pretty easily copy the source over compile and go. And things just work (barring the occasional endian issue that happens when taking software between architectures that are big vs little vs bi endian)

 

Now theres always talk of a new cheap x86 based SoC such as the Edison, or other Arm based SBC like the odriod, or the CHIP, but I have yet to see one that is as readily available as the Pi at the $35 with the myriad of addons and software already running on it.

(Yesterday I installed FreeSWITCH on a Pi and it took about 45 minutes?  there was 1 issues with Faxing that we are looking into and its probably an endian issue) 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Colton Conor
Sent: Friday, June 3, 2016 7:59 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] SwitchPi

 

May I ask the question of why make something customer just to run on the Rasberry Pi? There are X86 SOC coming to the market that will be close to the cost of the Pi, and be much more powerful. Plus, you can run regular x86 operating systems. 

 

On Thu, Jun 2, 2016 at 12:11 PM, Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org> > wrote:

The custom boards allow for hardware DTMF keypad interface, hook switch and a few other things like LCD for CLID on old school phones with that provision.

 

There will be official Raspian packages sooner or later? theres a know issue with the cowbuilder that?s blocking this at this time, we?re working on a work around? 

 

From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>  [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> ] On Behalf Of William Scott
Sent: Wednesday, June 1, 2016 6:52 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Subject: Re: [Freeswitch-users] SwitchPi

 

Hi

 

Yes, I have FusionPBX/freeSwitch running on a RPi3. Will the ARM build ever become "Official"?

 

I've watched a 'tube video or two. What is the functionality of the custom boards?

  <https://mailtrack.io/trace/mail/d1a10f8bcbe53cd87724e18dfebb1cfaa57d5246383145.png> 


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

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

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

 

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From bipin at xbipin.com  Fri Jun  3 19:51:32 2016
From: bipin at xbipin.com (Bipin Patel)
Date: Fri, 3 Jun 2016 19:51:32 +0400
Subject: [Freeswitch-users] SwitchPi
In-Reply-To: <095c01d1bda2$2159f430$640ddc90$@freeswitch.org>
References: <CABV9BUu8ygpbbGLzxmzG70NsiS7dj5D4WRkNaQH6Hzex_Q38yg@mail.gmail.com>
	<00bb01d1bc20$ab399cb0$01acd610$@freeswitch.org>
	<CABV9BUuns9Z+-KYJvj8LeRrDUkKBnL-Dh46dXJ-Y2kDFent2Rg@mail.gmail.com>
	<04ea01d1bcf1$c1a373f0$44ea5bd0$@freeswitch.org>
	<CAMDdSzMj5-GU1M1930ueDFGMDiZPTwUqgVFr=Gj2ukGxT2-W2w@mail.gmail.com>
	<095c01d1bda2$2159f430$640ddc90$@freeswitch.org>
Message-ID: <7fe71dc2-bbc7-3197-96c0-5e1ef3512cc8@xbipin.com>

just to add, last time i compiled FS on rpi3 using a samsung evo plus 
card, it took like 25mins or so at most not to mention its getting 
powerful every year, now only if the codecs could make use of the neon 
in the processor.


Regards,
Bipin


------------------------------------------------------------------------
-------- Original Message --------
Subject: Re: [Freeswitch-users] SwitchPi
From: Ken Rice <krice at freeswitch.org>
To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
Date: 6/3/2016, 6:13:39 PM
>
> Why the Pi you ask? Why custom hardware to go with it?
>
> Generic X86 Hardware is not the end all be all. While the X86 based 
> CPUs means that there can be less portability issues than say 
> something like ARM or SPARC or other architectures, the PI still 
> typically offers several other capabilities that may not be offered on 
> the other Single Board Computers at a $35 price point.
>
> Then there is the well documented pile of GPIO ports that make it 
> quite easy to drive things like custom keypads, servos, steppers, etc 
> etc etc?  this means you can spin a small custom board (in my case its 
> about 1.5 square inches) that has the specific additional hardware for 
> adding an additional 24 GPIOs giving room for a combination of 
> something like 36 buttons worth of keypads and 12 other dry contact 
> inputs and outputs (if you use 12 of the GPIO pins to drive a keypad 
> array you can get 36 buttons. Basic formula  for max number of 
> potential keys   (n/2)^2 where n is the number of GPIO pins used in 
> the keypad array.
>
> Now mix that with that fact that ARM is not a second class Linux 
> citizen anymore, we can pretty easily copy the source over compile and 
> go. And things just work (barring the occasional endian issue that 
> happens when taking software between architectures that are big vs 
> little vs bi endian)
>
> Now theres always talk of a new cheap x86 based SoC such as the 
> Edison, or other Arm based SBC like the odriod, or the CHIP, but I 
> have yet to see one that is as readily available as the Pi at the $35 
> with the myriad of addons and software already running on it.
>
> (Yesterday I installed FreeSWITCH on a Pi and it took about 45 
> minutes?  there was 1 issues with Faxing that we are looking into and 
> its probably an endian issue)
>
> *From:*freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
> *Colton Conor
> *Sent:* Friday, June 3, 2016 7:59 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] SwitchPi
>
> May I ask the question of why make something customer just to run on 
> the Rasberry Pi? There are X86 SOC coming to the market that will be 
> close to the cost of the Pi, and be much more powerful. Plus, you can 
> run regular x86 operating systems.
>
> On Thu, Jun 2, 2016 at 12:11 PM, Ken Rice <krice at freeswitch.org 
> <mailto:krice at freeswitch.org>> wrote:
>
>     The custom boards allow for hardware DTMF keypad interface, hook
>     switch and a few other things like LCD for CLID on old school
>     phones with that provision.
>
>     There will be official Raspian packages sooner or later? theres a
>     know issue with the cowbuilder that?s blocking this at this time,
>     we?re working on a work around?
>
>     *From:*freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>
>     [mailto:freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>] *On Behalf
>     Of *William Scott
>     *Sent:* Wednesday, June 1, 2016 6:52 PM
>     *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>
>     *Subject:* Re: [Freeswitch-users] SwitchPi
>
>     Hi
>
>     Yes, I have FusionPBX/freeSwitch running on a RPi3. Will the ARM
>     build ever become "Official"?
>
>     I've watched a 'tube video or two. What is the functionality of
>     the custom boards?
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From jack at livematch.com  Sat Jun  4 00:40:03 2016
From: jack at livematch.com (Jack Loranger)
Date: Fri, 3 Jun 2016 13:40:03 -0700
Subject: [Freeswitch-users] Windows 2012 Video
Message-ID: <fa787ffb-56af-b7b9-b2e3-a91f3b16bd57@livematch.com>

Has anyone been successful serving video from windows 2012?

This is the error I receive:
2016-06-03 13:32:01.472079 [ERR] mod_conference.c:2835 video-mode 
invalid, only valid setting is 'passthrough' due to no video capabilities

Thanks,
  Jack



From mike at jerris.com  Sat Jun  4 00:47:53 2016
From: mike at jerris.com (Michael Jerris)
Date: Fri, 3 Jun 2016 16:47:53 -0400
Subject: [Freeswitch-users] Windows 2012 Video
In-Reply-To: <fa787ffb-56af-b7b9-b2e3-a91f3b16bd57@livematch.com>
References: <fa787ffb-56af-b7b9-b2e3-a91f3b16bd57@livematch.com>
Message-ID: <60FFED11-029F-4FA0-A6A2-E9A9875D411D@jerris.com>

I still have not added the video dependencies to the windows build.  We will need to at the very least get libvpx and libyuv and libpng building for windows and some defines moved around for that feature to work.

> On Jun 3, 2016, at 4:40 PM, Jack Loranger <jack at livematch.com> wrote:
> 
> Has anyone been successful serving video from windows 2012?
> 
> This is the error I receive:
> 2016-06-03 13:32:01.472079 [ERR] mod_conference.c:2835 video-mode 
> invalid, only valid setting is 'passthrough' due to no video capabilities
> 
> Thanks,
>  Jack
> 


From bipin at xbipin.com  Sat Jun  4 00:48:34 2016
From: bipin at xbipin.com (Bipin Patel)
Date: Sat, 04 Jun 2016 00:48:34 +0400
Subject: [Freeswitch-users] Windows 2012 Video
In-Reply-To: <fa787ffb-56af-b7b9-b2e3-a91f3b16bd57@livematch.com>
References: <fa787ffb-56af-b7b9-b2e3-a91f3b16bd57@livematch.com>
Message-ID: <15518064cd0.2787.b07ebdf329620b8089087c7205b03f01@xbipin.com>

I was attempting to do the same so good to know I'm not alone. I was told 
it might not work due to some libs which can't be redistributed etc, not 
sure of which ones but I guess if some one puts the effort it could work.




On June 4, 2016 12:41:53 AM Jack Loranger <jack at livematch.com> wrote:

> Has anyone been successful serving video from windows 2012?
>
> This is the error I receive:
> 2016-06-03 13:32:01.472079 [ERR] mod_conference.c:2835 video-mode
> invalid, only valid setting is 'passthrough' due to no video capabilities
>
> Thanks,
>   Jack
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From mike at jerris.com  Sat Jun  4 00:54:35 2016
From: mike at jerris.com (Michael Jerris)
Date: Fri, 3 Jun 2016 16:54:35 -0400
Subject: [Freeswitch-users] Windows 2012 Video
In-Reply-To: <15518064cd0.2787.b07ebdf329620b8089087c7205b03f01@xbipin.com>
References: <fa787ffb-56af-b7b9-b2e3-a91f3b16bd57@livematch.com>
	<15518064cd0.2787.b07ebdf329620b8089087c7205b03f01@xbipin.com>
Message-ID: <2DE5A57C-7DA2-4AC9-B9B7-E9ACFD4A5638@jerris.com>

The libs that can't be distributed are the h264 libs.  This does not mean we can't add build files to build those things, but they will never be in our binary versions.  It's not a matter of might not, for 100% sure, the video code and dependencies are disabled in the windows build, and the windows build needs to be updated to include these new dependencies and to turn on the video code.  


> On Jun 3, 2016, at 4:48 PM, Bipin Patel <bipin at xbipin.com> wrote:
> 
> I was attempting to do the same so good to know I'm not alone. I was told 
> it might not work due to some libs which can't be redistributed etc, not 
> sure of which ones but I guess if some one puts the effort it could work.
> 
> 
> 
> 
> On June 4, 2016 12:41:53 AM Jack Loranger <jack at livematch.com> wrote:
> 
>> Has anyone been successful serving video from windows 2012?
>> 
>> This is the error I receive:
>> 2016-06-03 13:32:01.472079 [ERR] mod_conference.c:2835 video-mode
>> invalid, only valid setting is 'passthrough' due to no video capabilities
>> 
>> Thanks,
>>  Jack
>> 



From jack at livematch.com  Sat Jun  4 02:02:56 2016
From: jack at livematch.com (Jack Loranger)
Date: Fri, 3 Jun 2016 15:02:56 -0700
Subject: [Freeswitch-users] Windows 2012 Video
In-Reply-To: <2DE5A57C-7DA2-4AC9-B9B7-E9ACFD4A5638@jerris.com>
References: <fa787ffb-56af-b7b9-b2e3-a91f3b16bd57@livematch.com>
	<15518064cd0.2787.b07ebdf329620b8089087c7205b03f01@xbipin.com>
	<2DE5A57C-7DA2-4AC9-B9B7-E9ACFD4A5638@jerris.com>
Message-ID: <dd4aa5e5-dc7b-bed3-de97-209245d87f37@livematch.com>

Thanks for the comprehensive answer.

On 6/3/2016 1:54 PM, Michael Jerris wrote:
> The libs that can't be distributed are the h264 libs.  This does not mean we can't add build files to build those things, but they will never be in our binary versions.  It's not a matter of might not, for 100% sure, the video code and dependencies are disabled in the windows build, and the windows build needs to be updated to include these new dependencies and to turn on the video code.
>
>
>> On Jun 3, 2016, at 4:48 PM, Bipin Patel <bipin at xbipin.com> wrote:
>>
>> I was attempting to do the same so good to know I'm not alone. I was told
>> it might not work due to some libs which can't be redistributed etc, not
>> sure of which ones but I guess if some one puts the effort it could work.
>>
>>
>>
>>
>> On June 4, 2016 12:41:53 AM Jack Loranger <jack at livematch.com> wrote:
>>
>>> Has anyone been successful serving video from windows 2012?
>>>
>>> This is the error I receive:
>>> 2016-06-03 13:32:01.472079 [ERR] mod_conference.c:2835 video-mode
>>> invalid, only valid setting is 'passthrough' due to no video capabilities
>>>
>>> Thanks,
>>>   Jack
>>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> -----
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2016.0.7639 / Virus Database: 4591/12354 - Release Date: 06/03/16
>
>




From naveen32india at gmail.com  Sat Jun  4 04:01:43 2016
From: naveen32india at gmail.com (Naveen Tamanam)
Date: Sat, 4 Jun 2016 05:31:43 +0530
Subject: [Freeswitch-users] Unable to install module mod_flite on CentOS
In-Reply-To: <0406510B-6297-4834-8030-B4E4B7B5278E@jerris.com>
References: <CABsWbRbrgh3g9yS9tJF87ARg0nVv3qi=1y-FWSjaT=sa4ZSgdw@mail.gmail.com>
	<ed34459e2c784a6dbf5d90c57f73b0e7@nysolutions.com>
	<CABsWbRauppe2KKQzEv92+sb8HrbQANaMVG7=AvXoDp_XnMGJ0Q@mail.gmail.com>
	<0406510B-6297-4834-8030-B4E4B7B5278E@jerris.com>
Message-ID: <CABsWbRYF69L6KNjPoYCYNyGDvXRp+HRxJg9NDBsHyC720VVFRQ@mail.gmail.com>

Hi Jerris,
Yes I run the ./configure   after I install this module

On Thu, Jun 2, 2016 at 10:19 PM, Michael Jerris <mike at jerris.com> wrote:

> This shoud be the right one.  Did you run freeswitch configure after you
> installed this?
>
> On Jun 1, 2016, at 7:19 PM, Naveen Tamanam <naveen32india at gmail.com>
> wrote:
>
> Yes, I have installed libflite-dev from
> https://freeswitch.org/stash/scm/sd/libflite.git
> Manually compiled it and installed it.
>
> On Thu, Jun 2, 2016 at 4:41 AM, Moishe Grunstein <max at nysolutions.com>
> wrote:
>
>> Did you install libflite-dev?
>>
>>
>>
>> Thanks,
>>
>>
>>
>> Moishe Grunstein
>>
>> Tornado Computer Systems, Inc.
>>
>> 212.400.7650 888.IPPBX.US <http://888.ippbx.us/>
>> *Service Request Email: support at nysolutions.com <support at nysolutions.com>
>> *
>>
>> <image001.jpg> <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-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Naveen
>> Tamanam
>> *Sent:* Wednesday, June 1, 2016 5:44 PM
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* [Freeswitch-users] Unable to install module mod_flite on
>> CentOS
>>
>>
>>
>> Hi All,
>>
>> I'm trying to compile freeswitch with module mod_flite. I have downloaded
>> the module from
>>      https://freeswitch.org/stash/scm/sd/libflite.git
>>
>> Compiled it and installed successfully. Then trying to compile
>> freeswitch with mod_flite enabled, still I'm getting following error (for
>> make),
>>
>> making all mod_flite
>> make[4]: Entering directory
>> `/usr/local/src/freeswitch/src/mod/asr_tts/mod_flite'
>> Makefile:886: *** You must install libflite-dev to build mod_flite.  Stop.
>> make[4]: Leaving directory
>> `/usr/local/src/freeswitch/src/mod/asr_tts/mod_flite'
>> make[3]: *** [mod_flite-all] Error 1
>> make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod'
>> make[2]: *** [all-recursive] Error 1
>> make[2]: Leaving directory `/usr/local/src/freeswitch/src'
>> make[1]: *** [all-recursive] Error 1
>> make[1]: Leaving directory `/usr/local/src/freeswitch'
>> make: *** [all] Error 2
>>
>> I'm using centOS 7
>>
>> freeswitch version --  1.6.2
>>
>>
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Thanks & Regards,
Naveen Tamanam
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From ssinyagin at gmail.com  Sat Jun  4 04:11:42 2016
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sat, 4 Jun 2016 02:11:42 +0200
Subject: [Freeswitch-users] SwitchPi
In-Reply-To: <095c01d1bda2$2159f430$640ddc90$@freeswitch.org>
References: <CABV9BUu8ygpbbGLzxmzG70NsiS7dj5D4WRkNaQH6Hzex_Q38yg@mail.gmail.com>
	<00bb01d1bc20$ab399cb0$01acd610$@freeswitch.org>
	<CABV9BUuns9Z+-KYJvj8LeRrDUkKBnL-Dh46dXJ-Y2kDFent2Rg@mail.gmail.com>
	<04ea01d1bcf1$c1a373f0$44ea5bd0$@freeswitch.org>
	<CAMDdSzMj5-GU1M1930ueDFGMDiZPTwUqgVFr=Gj2ukGxT2-W2w@mail.gmail.com>
	<095c01d1bda2$2159f430$640ddc90$@freeswitch.org>
Message-ID: <CACOYK=gucwrvYqJRLCP8x9w6EQg2L8RwqG8SZV2y=u+bYLjPiA@mail.gmail.com>

Ken, you probably hit this bug with faxing:
https://freeswitch.org/jira/browse/FS-8651

we're still waiting for Steve's feedback on that one.





On Fri, Jun 3, 2016 at 4:13 PM, Ken Rice <krice at freeswitch.org> wrote:

> Why the Pi you ask? Why custom hardware to go with it?
>
>
>
> Generic X86 Hardware is not the end all be all. While the X86 based CPUs
> means that there can be less portability issues than say something like ARM
> or SPARC or other architectures, the PI still typically offers several
> other capabilities that may not be offered on the other Single Board
> Computers at a $35 price point.
>
>
>
> Then there is the well documented pile of GPIO ports that make it quite
> easy to drive things like custom keypads, servos, steppers, etc etc etc?
>  this means you can spin a small custom board (in my case its about 1.5
> square inches) that has the specific additional hardware for adding an
> additional 24 GPIOs giving room for a combination of something like 36
> buttons worth of keypads and 12 other dry contact inputs and outputs (if
> you use 12 of the GPIO pins to drive a keypad array you can get 36 buttons.
> Basic formula  for max number of potential keys   (n/2)^2 where n is the
> number of GPIO pins used in the keypad array.
>
>
>
> Now mix that with that fact that ARM is not a second class Linux citizen
> anymore, we can pretty easily copy the source over compile and go. And
> things just work (barring the occasional endian issue that happens when
> taking software between architectures that are big vs little vs bi endian)
>
>
>
> Now theres always talk of a new cheap x86 based SoC such as the Edison, or
> other Arm based SBC like the odriod, or the CHIP, but I have yet to see one
> that is as readily available as the Pi at the $35 with the myriad of addons
> and software already running on it.
>
> (Yesterday I installed FreeSWITCH on a Pi and it took about 45 minutes?
> there was 1 issues with Faxing that we are looking into and its probably an
> endian issue)
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Colton Conor
> *Sent:* Friday, June 3, 2016 7:59 AM
>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] SwitchPi
>
>
>
> May I ask the question of why make something customer just to run on the
> Rasberry Pi? There are X86 SOC coming to the market that will be close to
> the cost of the Pi, and be much more powerful. Plus, you can run regular
> x86 operating systems.
>
>
>
> On Thu, Jun 2, 2016 at 12:11 PM, Ken Rice <krice at freeswitch.org> wrote:
>
> The custom boards allow for hardware DTMF keypad interface, hook switch
> and a few other things like LCD for CLID on old school phones with that
> provision.
>
>
>
> There will be official Raspian packages sooner or later? theres a know
> issue with the cowbuilder that?s blocking this at this time, we?re working
> on a work around?
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *William
> Scott
> *Sent:* Wednesday, June 1, 2016 6:52 PM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] SwitchPi
>
>
>
> Hi
>
>
>
> Yes, I have FusionPBX/freeSwitch running on a RPi3. Will the ARM build
> ever become "Official"?
>
>
>
> I've watched a 'tube video or two. What is the functionality of the custom
> boards?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From krice at freeswitch.org  Sat Jun  4 04:54:02 2016
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 3 Jun 2016 19:54:02 -0500
Subject: [Freeswitch-users] SwitchPi
In-Reply-To: <CACOYK=gucwrvYqJRLCP8x9w6EQg2L8RwqG8SZV2y=u+bYLjPiA@mail.gmail.com>
References: <CABV9BUu8ygpbbGLzxmzG70NsiS7dj5D4WRkNaQH6Hzex_Q38yg@mail.gmail.com>
	<00bb01d1bc20$ab399cb0$01acd610$@freeswitch.org>
	<CABV9BUuns9Z+-KYJvj8LeRrDUkKBnL-Dh46dXJ-Y2kDFent2Rg@mail.gmail.com>
	<04ea01d1bcf1$c1a373f0$44ea5bd0$@freeswitch.org>
	<CAMDdSzMj5-GU1M1930ueDFGMDiZPTwUqgVFr=Gj2ukGxT2-W2w@mail.gmail.com>
	<095c01d1bda2$2159f430$640ddc90$@freeswitch.org>
	<CACOYK=gucwrvYqJRLCP8x9w6EQg2L8RwqG8SZV2y=u+bYLjPiA@mail.gmail.com>
Message-ID: <C3945830-AF48-4982-8B66-ABE04C6D0CB6@freeswitch.org>

I asked steve about this and yes i have duplicated it. Looks like possibly an endian issue

Sent from my iPhone

> On Jun 3, 2016, at 7:11 PM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
> 
> Ken, you probably hit this bug with faxing:
> https://freeswitch.org/jira/browse/FS-8651
> 
> we're still waiting for Steve's feedback on that one.
> 
> 
> 
> 
> 
>> On Fri, Jun 3, 2016 at 4:13 PM, Ken Rice <krice at freeswitch.org> wrote:
>> Why the Pi you ask? Why custom hardware to go with it?
>> 
>>  
>> 
>> Generic X86 Hardware is not the end all be all. While the X86 based CPUs means that there can be less portability issues than say something like ARM or SPARC or other architectures, the PI still typically offers several other capabilities that may not be offered on the other Single Board Computers at a $35 price point.
>> 
>>  
>> 
>> Then there is the well documented pile of GPIO ports that make it quite easy to drive things like custom keypads, servos, steppers, etc etc etc?  this means you can spin a small custom board (in my case its about 1.5 square inches) that has the specific additional hardware for adding an additional 24 GPIOs giving room for a combination of something like 36 buttons worth of keypads and 12 other dry contact inputs and outputs (if you use 12 of the GPIO pins to drive a keypad array you can get 36 buttons. Basic formula  for max number of potential keys   (n/2)^2 where n is the number of GPIO pins used in the keypad array.
>> 
>>  
>> 
>> Now mix that with that fact that ARM is not a second class Linux citizen anymore, we can pretty easily copy the source over compile and go. And things just work (barring the occasional endian issue that happens when taking software between architectures that are big vs little vs bi endian)
>> 
>>  
>> 
>> Now theres always talk of a new cheap x86 based SoC such as the Edison, or other Arm based SBC like the odriod, or the CHIP, but I have yet to see one that is as readily available as the Pi at the $35 with the myriad of addons and software already running on it.
>> 
>> (Yesterday I installed FreeSWITCH on a Pi and it took about 45 minutes?  there was 1 issues with Faxing that we are looking into and its probably an endian issue)
>> 
>>  
>> 
>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Colton Conor
>> Sent: Friday, June 3, 2016 7:59 AM
>> 
>> 
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Subject: Re: [Freeswitch-users] SwitchPi
>>  
>> 
>> May I ask the question of why make something customer just to run on the Rasberry Pi? There are X86 SOC coming to the market that will be close to the cost of the Pi, and be much more powerful. Plus, you can run regular x86 operating systems. 
>> 
>>  
>> 
>> On Thu, Jun 2, 2016 at 12:11 PM, Ken Rice <krice at freeswitch.org> wrote:
>> 
>> The custom boards allow for hardware DTMF keypad interface, hook switch and a few other things like LCD for CLID on old school phones with that provision.
>> 
>>  
>> 
>> There will be official Raspian packages sooner or later? theres a know issue with the cowbuilder that?s blocking this at this time, we?re working on a work around?
>> 
>>  
>> 
>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William Scott
>> Sent: Wednesday, June 1, 2016 6:52 PM
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Subject: Re: [Freeswitch-users] SwitchPi
>> 
>>  
>> 
>> Hi
>> 
>>  
>> 
>> Yes, I have FusionPBX/freeSwitch running on a RPi3. Will the ARM build ever become "Official"?
>> 
>>  
>> 
>> I've watched a 'tube video or two. What is the functionality of the custom boards?
>> 
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>>  
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From naveen32india at gmail.com  Sat Jun  4 04:56:04 2016
From: naveen32india at gmail.com (Naveen Tamanam)
Date: Sat, 4 Jun 2016 06:26:04 +0530
Subject: [Freeswitch-users] Unable to install module mod_flite on CentOS
In-Reply-To: <CABsWbRYF69L6KNjPoYCYNyGDvXRp+HRxJg9NDBsHyC720VVFRQ@mail.gmail.com>
References: <CABsWbRbrgh3g9yS9tJF87ARg0nVv3qi=1y-FWSjaT=sa4ZSgdw@mail.gmail.com>
	<ed34459e2c784a6dbf5d90c57f73b0e7@nysolutions.com>
	<CABsWbRauppe2KKQzEv92+sb8HrbQANaMVG7=AvXoDp_XnMGJ0Q@mail.gmail.com>
	<0406510B-6297-4834-8030-B4E4B7B5278E@jerris.com>
	<CABsWbRYF69L6KNjPoYCYNyGDvXRp+HRxJg9NDBsHyC720VVFRQ@mail.gmail.com>
Message-ID: <CABsWbRb_0+6d+HvfU2WRFO=ervTfKrgkugPFZg4NFtLHBaG1Bg@mail.gmail.com>

Thank you very much it's working.  I have tried
https://okay.com.mx/en/blog/rpm-repositories-for-centos-6-and-7.html

On Sat, Jun 4, 2016 at 5:31 AM, Naveen Tamanam <naveen32india at gmail.com>
wrote:

> Hi Jerris,
> Yes I run the ./configure   after I install this module
>
> On Thu, Jun 2, 2016 at 10:19 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> This shoud be the right one.  Did you run freeswitch configure after you
>> installed this?
>>
>> On Jun 1, 2016, at 7:19 PM, Naveen Tamanam <naveen32india at gmail.com>
>> wrote:
>>
>> Yes, I have installed libflite-dev from
>> https://freeswitch.org/stash/scm/sd/libflite.git
>> Manually compiled it and installed it.
>>
>> On Thu, Jun 2, 2016 at 4:41 AM, Moishe Grunstein <max at nysolutions.com>
>> wrote:
>>
>>> Did you install libflite-dev?
>>>
>>>
>>>
>>> Thanks,
>>>
>>>
>>>
>>> Moishe Grunstein
>>>
>>> Tornado Computer Systems, Inc.
>>>
>>> 212.400.7650 888.IPPBX.US <http://888.ippbx.us/>
>>> *Service Request Email: support at nysolutions.com
>>> <support at nysolutions.com> *
>>>
>>> <image001.jpg> <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-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Naveen
>>> Tamanam
>>> *Sent:* Wednesday, June 1, 2016 5:44 PM
>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> *Subject:* [Freeswitch-users] Unable to install module mod_flite on
>>> CentOS
>>>
>>>
>>>
>>> Hi All,
>>>
>>> I'm trying to compile freeswitch with module mod_flite. I have
>>> downloaded the module from
>>>      https://freeswitch.org/stash/scm/sd/libflite.git
>>>
>>> Compiled it and installed successfully. Then trying to compile
>>> freeswitch with mod_flite enabled, still I'm getting following error (for
>>> make),
>>>
>>> making all mod_flite
>>> make[4]: Entering directory
>>> `/usr/local/src/freeswitch/src/mod/asr_tts/mod_flite'
>>> Makefile:886: *** You must install libflite-dev to build mod_flite.
>>> Stop.
>>> make[4]: Leaving directory
>>> `/usr/local/src/freeswitch/src/mod/asr_tts/mod_flite'
>>> make[3]: *** [mod_flite-all] Error 1
>>> make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod'
>>> make[2]: *** [all-recursive] Error 1
>>> make[2]: Leaving directory `/usr/local/src/freeswitch/src'
>>> make[1]: *** [all-recursive] Error 1
>>> make[1]: Leaving directory `/usr/local/src/freeswitch'
>>> make: *** [all] Error 2
>>>
>>> I'm using centOS 7
>>>
>>> freeswitch version --  1.6.2
>>>
>>>
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Thanks & Regards,
> Naveen Tamanam
>



-- 
Thanks & Regards,
Naveen Tamanam
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From 35633 at heb.be  Sat Jun  4 11:43:31 2016
From: 35633 at heb.be (Nduwayezu, Joselyne)
Date: Sat, 4 Jun 2016 09:43:31 +0200
Subject: [Freeswitch-users] Git error
Message-ID: <CAFSzUtaM6cR+R6KnOA8NuU63rtPKqytAgnQ7MyFf1i4346rL+A@mail.gmail.com>

Hello,
I'm trying to have clone this one:

*1) git clone git://git.freeswitch.org/freeswitch-contrib.git
<http://git.freeswitch.org/freeswitch-contrib.git>*

but i have the following error

"fatal: unable to connecte to git.freeswitch.org

git.freeswitch.org[0: 209.105.235.6]: erno=Connection refused

git.freeswitch.og[1: 2607:f348:1021::6]: erno=Network is unreachable

But when i ping git.freeswitch.org or ping 209.105.235.6, the ping success

2)

*wget http://music.incompetech.com/royalty-free/Parisian.mp3
<http://music.incompetech.com/royalty-free/Parisian.mp3> *

the error is:

HTTP request sent, awaiting response ....  404 Not Found

Any ideas? Thanks

NDUWAYEZU Joselyne

-- 
Haute ?cole de Bruxelles
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From 35633 at heb.be  Sat Jun  4 12:28:09 2016
From: 35633 at heb.be (Nduwayezu, Joselyne)
Date: Sat, 4 Jun 2016 10:28:09 +0200
Subject: [Freeswitch-users] failed to install mod_flite
Message-ID: <CAFSzUtZGVocBhJspdaO4CPQMjOHcypM_t3MpuMgU9nXwPR0HWQ@mail.gmail.com>

I'm trying to compile freeswitch with module mod_flite. I have downloaded
the module from
     https://freeswitch.org/stash/scm/sd/libflite.git
Compiled it and installed successfully. Then trying to compile  freeswitch
with mod_flite enabled, still I'm getting following error (for make),

making all mod_flite
make[4]: Entering directory
`/usr/local/src/freeswitch/src/mod/asr_tts/mod_flite'
Makefile:886: *** You must install libflite-dev to build mod_flite.  Stop.
make[4]: Leaving directory
`/usr/local/src/freeswitch/src/mod/asr_tts/mod_flite'
make[3]: *** [mod_flite-all] Error 1
make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod'
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory `/usr/local/src/freeswitch/src'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/usr/local/src/freeswitch'
make: *** [all] Error 2

I'm using ubuntu 15.10

NDUWAYEZU Joselyne

-- 
Haute ?cole de Bruxelles
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From 35633 at heb.be  Sat Jun  4 12:36:47 2016
From: 35633 at heb.be (Nduwayezu, Joselyne)
Date: Sat, 4 Jun 2016 10:36:47 +0200
Subject: [Freeswitch-users] x-lite and freeswitch
Message-ID: <CAFSzUtYMOKBw1zFQLcrWb9KtRUvMLDut4xh4myk0dQW4nTdL5Q@mail.gmail.com>

I would like to register x-lite with Freeswitch.
But it fails to register when "register with domain and receive calls" is
checked.

What may be the reason of that ?

NDUWAYEZU Joselyne

-- 
Haute ?cole de Bruxelles
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From 35633 at heb.be  Sat Jun  4 14:37:32 2016
From: 35633 at heb.be (Nduwayezu, Joselyne)
Date: Sat, 4 Jun 2016 12:37:32 +0200
Subject: [Freeswitch-users] Problem with autogen syntax error
Message-ID: <CAFSzUtbSjAP98oqKFh4V3ErvPuN3QckO6i9aQ=inAA98UNLRKA@mail.gmail.com>

Hello i would like to use tts_commandline engine.
When i execute the command sh ./autogen.sh, i havee these errors:


root at back-2:/usr/local/src/freeswitch/freeswitch-contrib/grmt/svox
pico/svox/pico# sh ./autogen.sh
aclocal: warning: autoconf input should be named 'configure.ac', not '
configure.in'
aclocal: error: couldn't open directory 'm4': No such file or directory
libtoolize: putting auxiliary files in `.'.
libtoolize: linking file `./ltmain.sh'
libtoolize: putting macros in AC_CONFIG_MACRO_DIR, `m4'.
libtoolize: linking file `m4/libtool.m4'
libtoolize: linking file `m4/ltoptions.m4'
libtoolize: linking file `m4/ltsugar.m4'
libtoolize: linking file `m4/ltversion.m4'
libtoolize: linking file `m4/lt~obsolete.m4'
configure.in:7: error: possibly undefined macro: AM_INIT_AUTOMAKE
      If this token and others are legitimate, please use m4_pattern_allow.
      See the Autoconf documentation.
configure.in:11: error: possibly undefined macro: AC_PROG_LIBTOOL
automake: warning: autoconf input should be named 'configure.ac', not '
configure.in'
configure.in: error: no proper invocation of AM_INIT_AUTOMAKE was found.
configure.in: You should verify that configure.in invokes AM_INIT_AUTOMAKE,
configure.in: that aclocal.m4 is present in the top-level directory,
configure.in: and that aclocal.m4 was recently regenerated (using aclocal)
Makefile.am: installing './INSTALL'
Makefile.am: error: required file './NEWS' not found
Makefile.am: error: required file './README' not found
Makefile.am: error: required file './AUTHORS' not found
Makefile.am: error: required file './ChangeLog' not found
Makefile.am: installing './COPYING' using GNU General Public License v3 file
Makefile.am:     Consider adding the COPYING file to the version control
system
Makefile.am:     for your code, to avoid questions about which license your
project uses
automake: warning: autoconf input should be named 'configure.ac', not '
configure.in'
Makefile.am:5: error: Libtool library used but 'LIBTOOL' is undefined
Makefile.am:5:   The usual way to define 'LIBTOOL' is to add 'LT_INIT'
Makefile.am:5:   to 'configure.in' and run 'aclocal' and 'autoconf' again.
Makefile.am:5:   If 'LT_INIT' is in 'configure.in', make sure
Makefile.am:5:   its definition is in aclocal's search path.
Makefile.am:6: warning: source file 'lib/picoacph.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
automake: warning: possible forward-incompatibility.
automake: At least a source file is in a subdirectory, but the
'subdir-objects'
automake: automake option hasn't been enabled.  For now, the corresponding
output
automake: object file(s) will be placed in the top-level directory.
However,
automake: this behaviour will change in future Automake versions: they will
automake: unconditionally cause object files to be placed in the same
subdirectory
automake: of the corresponding sources.
automake: You are advised to start using 'subdir-objects' option throughout
your
automake: project, to avoid future incompatibilities.
Makefile.am:6: warning: source file 'lib/picoapi.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:6: warning: source file 'lib/picobase.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:6: warning: source file 'lib/picocep.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:6: warning: source file 'lib/picoctrl.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:6: warning: source file 'lib/picodata.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:6: warning: source file 'lib/picodbg.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:6: warning: source file 'lib/picoextapi.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:6: warning: source file 'lib/picofftsg.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:6: warning: source file 'lib/picokdbg.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:6: warning: source file 'lib/picokdt.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:6: warning: source file 'lib/picokfst.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:6: warning: source file 'lib/picoklex.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:6: warning: source file 'lib/picoknow.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:6: warning: source file 'lib/picokpdf.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:6: warning: source file 'lib/picokpr.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:6: warning: source file 'lib/picoktab.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:6: warning: source file 'lib/picoos.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:6: warning: source file 'lib/picopal.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:6: warning: source file 'lib/picopam.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:6: warning: source file 'lib/picopr.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:6: warning: source file 'lib/picorsrc.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:6: warning: source file 'lib/picosa.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:6: warning: source file 'lib/picosig.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:6: warning: source file 'lib/picosig2.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:6: warning: source file 'lib/picospho.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:6: warning: source file 'lib/picotok.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:6: warning: source file 'lib/picotrns.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:6: warning: source file 'lib/picowa.c' is in a subdirectory,
Makefile.am:6: but option 'subdir-objects' is disabled
Makefile.am:83: warning: source file 'bin/pico2wave.c' is in a subdirectory,
Makefile.am:83: but option 'subdir-objects' is disabled
Makefile.am: installing './depcomp'
/usr/share/automake-1.15/am/depend2.am: error: am__fastdepCC does not
appear in AM_CONDITIONAL
/usr/share/automake-1.15/am/depend2.am:   The usual way to define
'am__fastdepCC' is to add 'AC_PROG_CC'
/usr/share/automake-1.15/am/depend2.am:   to 'configure.in' and run
'aclocal' and 'autoconf' again
/usr/share/automake-1.15/am/depend2.am: error: AMDEP does not appear in
AM_CONDITIONAL
/usr/share/automake-1.15/am/depend2.am:   The usual way to define 'AMDEP'
is to add one of the compiler tests
/usr/share/automake-1.15/am/depend2.am:     AC_PROG_CC, AC_PROG_CXX,
AC_PROG_OBJC, AC_PROG_OBJCXX,
/usr/share/automake-1.15/am/depend2.am:     AM_PROG_AS, AM_PROG_GCJ,
AM_PROG_UPC
/usr/share/automake-1.15/am/depend2.am:   to 'configure.in' and run
'aclocal' and 'autoconf' again
Now run ./configure and then make.
root at back-2:/usr/local/src/freeswitch/freeswitch-contrib/grmt/svox
pico/svox/pico# ls
AndroidManifest.xml  autogen.sh  configure     COPYING  INSTALL  lib
m4           res  tests
Android.mk           bin         configure.in  depcomp  lang     ltmain.sh
Makefile.am  src  tts
root at back-2:/usr/local/src/freeswitch/freeswitch-contrib/grmt/svox

pico/svox/pico# ./configure
./configure: line 1776: syntax error near unexpected token `1.9'
./configure: line 1776: `AM_INIT_AUTOMAKE(1.9 foreign)'
root at back-2:/usr/local/src/freeswitch/freeswitch-contrib/grmt/svox
pico/svox/pico# ls
AndroidManifest.xml  autogen.sh  config.log  configure.in  depcomp  lang
ltmain.sh  Makefile.am  src    tts
Android.mk           bin         configure   COPYING       INSTALL  lib
m4         res          tests
root at back-2:/usr/local/src/freeswitch/freeswitch-contrib/grmt/svox
pico/svox/pico# ./configure
./configure: line 1776: syntax error near unexpected token `1.9'
./configure: line 1776: `AM_INIT_AUTOMAKE(1.9 foreign)'
root at back-2:/usr/local/src/freeswitch/freeswitch-contrib/grmt/svox
pico/svox/pico# make && make install
make: *** No targets specified and no makefile found.  Stop.
root at back-2:/usr/local/src/freeswitch/freeswitch-contrib/grmt/svox
pico/svox/pico#

Any help please?

NDUWAYEZU Joselyne

-- 
Haute ?cole de Bruxelles
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From lists at kavun.ch  Sat Jun  4 15:44:39 2016
From: lists at kavun.ch (Emrah)
Date: Sat, 4 Jun 2016 13:44:39 +0200
Subject: [Freeswitch-users] Presence with shared database
In-Reply-To: <55C8832F.2030804@deanconnect.nl>
References: <CAMJTtGU3Hoi5KyduzADMV4q1G-FH44yP+rpXjPrW_KNT18P8HQ@mail.gmail.com>
	<55C8832F.2030804@deanconnect.nl>
Message-ID: <78A74FC5-FB22-4CB4-8261-FA5992E1206B@kavun.ch>

Hi there,
Has there been any changes on this end?
I am trying to store presence data in PGSQL in order to distribute it among a cluster of FS servers.
Is this achievable now?

Best,
Emrah
> On Aug 10, 2015, at 12:55 PM, Leo Noordergraaf <leo.noordergraaf at deanconnect.nl> wrote:
> 
> Hi Adam,
> 
> We ran into the same issue and created ticket
> https://freeswitch.org/jira/browse/FS-7485 <https://freeswitch.org/jira/browse/FS-7485> for it.
> For some reason it has been closed but the issue still remains open.
> 
> Like you we changed the select en update statements but it would be nice
> if it was solved by the freeswitch team.
> 
> regards,
> Leo Noordergraaf
> 
> On 08/10/15 12:08, Adam Ku?mirek wrote:
>> Hello,
>> 
>> Yes we enabled manage-presence.
>> According to documentation dbname is used to share presence between
>> profiles.
>> 
>> What we want to achieve is to share presence between hosts.
>> 
>> We investigated some presence source code and found that there are sql
>> queries that ask for specific FS host. We modified SELECT statements and
>> UPDATES of version parameter, so they have no host in WHERE clause.
>> After mod_sofia recompilation everything works fine.
>> 
>> It seems that the only way to share presence between hosts is to set the
>> same hostname in switch.xml for every FS node, but we cannot do this
>> because we need host information for other services.
>> 
>> I wonder why FS checks whether the subscription was made on a host that
>> sets up the connection. Maybe this is a bug in the concept and it should
>> be removed.
>> 
>> Regards
>> Adam Kusmirek
>> 
>> 
>> 
>>        ---------- Wiadomo?? przekazana dalej ----------
>> 
>>    From: "?talo Rossi" <italorossib at gmail.com
>>    <mailto:italorossib at gmail.com <mailto:italorossib at gmail.com>>>
>>    To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>
>>    <mailto:freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>>
>>    Cc: 
>>    Date: Fri, 7 Aug 2015 22:42:37 -0500
>>    Subject: Re: [Freeswitch-users] Presence with shared database
>>    Hard to say without more info, where's your configs?
>> 
>>    Did you enabled the manage-presence in your profile? and set the dbname?
>> 
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: 
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
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From lists at kavun.ch  Sat Jun  4 15:47:18 2016
From: lists at kavun.ch (Emrah)
Date: Sat, 4 Jun 2016 13:47:18 +0200
Subject: [Freeswitch-users] Sharing presence between FS boxes
In-Reply-To: <CAC-LwPMdES+0vwxT+1ohyyhLaRUkjJ7GCH5DEGGCkbwP7ewYQw@mail.gmail.com>
References: <1394821105.49776.YahooMailNeo@web171404.mail.ir2.yahoo.com>
	<1395044974.66092.YahooMailNeo@web171406.mail.ir2.yahoo.com>
	<CAC-LwPMdES+0vwxT+1ohyyhLaRUkjJ7GCH5DEGGCkbwP7ewYQw@mail.gmail.com>
Message-ID: <195141A1-A556-428F-890B-AFBDBBAB2A26@kavun.ch>

Hi there,
Any chance this can be accomplished without using the same hostname?

Best,
Emrah
> On Mar 17, 2014, at 12:59 PM, Vik Killa <vipkilla at gmail.com> wrote:
> 
> Set the hostname parameter in switch.conf.xml of both freeswitch boxes to the same name
> 
> 
> On Mon, Mar 17, 2014 at 4:29 AM, Chris B. Ware <chrisbware at yahoo.it <mailto:chrisbware at yahoo.it>> wrote:
> Anyone can help?
> Should I add all my domains on presence_hosts?
> 
> 
> 
> 
> Il Venerd? 14 Marzo 2014 19:18, Chris B. Ware <chrisbware at yahoo.it <mailto:chrisbware at yahoo.it>> ha scritto:
> Hi,
> 
> On advice by Brian and Anthony I'm writing to mailing list, after a jira (FS-6358).
> 
> If I have two Freeswitch servers, sharing the same DB as backend, how should I configure presence such that
> if a phone send SUBSCRIBE to box A, and call is received on box B, I get Notify from box B?
> 
> By now I've set manage_presence=true on both sip profiles (internal,external) and presence_hosts="_DISABLED_"
> as I read on wiki.
> 
> Of course dial_string contains presence_id=${dialed_user}@${dialed_domain}. Sip registrations are correctly shared
> between boxes, and even presence works sometimes, but is not  stable. 
> 
> Here my internal sip profiles config:
> 
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
>   <section name="configuration" description="Various Configuration">
>     <configuration name="sofia.conf" description="Sofia Endpoint">
>       <profiles>
>         <profile name="internal">
>           <domains>
>             <domain name="all" alias="true" parse="false"/>
>           </domains>
>           <settings>
>             <param name="debug" value="0"/>
>             <param name="sip-trace" value="no"/>
>             <param name="sip-capture" value="no"/>
>             <param name="watchdog-enabled" value="no"/>
>             <param name="watchdog-step-timeout" value="30000"/>
>             <param name="watchdog-event-timeout" value="30000"/>
>             <param name="log-auth-failures" value="true"/>
>             <param name="forward-unsolicited-mwi-notify" value="false"/>
>             <param name="context" value="default"/>
>             <param name="rfc2833-pt" value="101"/>
>             <param name="sip-port" value="$${internal_sip_port}"/>
>             <param name="dialplan" value="XML"/>
>             <param name="dtmf-duration" value="2000"/>
>             <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
>             <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
>             <param name="rtp-timer-name" value="soft"/>
>             <param name="rtp-ip" value="$${local_ip_v4}"/>
>             <param name="sip-ip" value="$${local_ip_v4}"/>
>             <param name="hold-music" value="$${hold_music}"/>
>             <param name="apply-nat-acl" value="nat.auto"/>
>             <param name="apply-inbound-acl" value="domains"/>
>             <param name="local-network-acl" value="localnet.auto"/>
>             <param name="record-path" value="$${recordings_dir}"/>
>             <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>             <param name="manage-presence" value="true"/>
>             <param name="inbound-codec-negotiation" value="generous"/>
>             <param name="tls" value="$${internal_ssl_enable}"/>
>             <param name="tls-bind-params" value="transport=tls"/>
>             <param name="tls-sip-port" value="$${internal_tls_port}"/>
>             <param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
>             <param name="tls-version" value="$${sip_tls_version}"/>
>             <param name="nonce-ttl" value="60"/>
>             <param name="auth-calls" value="$${internal_auth_calls}"/>
>             <param name="inbound-reg-force-matching-username" value="true"/>
>             <param name="auth-all-packets" value="false"/>
>             <param name="ext-rtp-ip" value="auto-nat"/>
>             <param name="ext-sip-ip" value="auto-nat"/>
>             <param name="rtp-timeout-sec" value="300"/>
>             <param name="rtp-hold-timeout-sec" value="1800"/>
>             <param name="presence-hosts" value="_DISABLED_"/>
>             <param name="odbc-dsn" value="systemDB:freeswitch:pazzword"/>
>             <param name="challenge-realm" value="auto_from"/>
>             <param name="outbound-proxy" value="sip::5060"/>
>           </settings>
>         </profile>
>       </profiles>
>     </configuration>
>   </section>
> </document>
> 
> 
> Can somebody help?
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
> http://www.cudatel.com <http://www.cudatel.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
> http://www.cudatel.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From krice at freeswitch.org  Sat Jun  4 17:30:03 2016
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 4 Jun 2016 08:30:03 -0500
Subject: [Freeswitch-users] Git error
In-Reply-To: <CAFSzUtaM6cR+R6KnOA8NuU63rtPKqytAgnQ7MyFf1i4346rL+A@mail.gmail.com>
References: <CAFSzUtaM6cR+R6KnOA8NuU63rtPKqytAgnQ7MyFf1i4346rL+A@mail.gmail.com>
Message-ID: <07E9BF9B-A19A-4B0F-8DE3-80C251F5413B@freeswitch.org>

Find some updated documentation on confluence at freeswitch.org

We dropped support for the git protocol ages ago. The git repo is accessible over https or ssh

Sent from my iPhone

> On Jun 4, 2016, at 2:43 AM, Nduwayezu, Joselyne <35633 at heb.be> wrote:
> 
> 
> Hello,
> I'm trying to have clone this one:
> 1) git clone git://git.freeswitch.org/freeswitch-contrib.git
> 
> but i have the following error
> 
> "fatal: unable to connecte to git.freeswitch.org
> 
> git.freeswitch.org[0: 209.105.235.6]: erno=Connection refused
> 
> git.freeswitch.og[1: 2607:f348:1021::6]: erno=Network is unreachable
> 
> But when i ping git.freeswitch.org or ping 209.105.235.6, the ping success
> 
> 2)
> 
> wget http://music.incompetech.com/royalty-free/Parisian.mp3
> 
> the error is:
> 
> HTTP request sent, awaiting response ....  404 Not Found
> 
> Any ideas? Thanks
> 
> 
> NDUWAYEZU Joselyne
> 
> Haute ?cole de Bruxelles
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From colin.morelli at gmail.com  Sat Jun  4 18:57:46 2016
From: colin.morelli at gmail.com (Colin Morelli)
Date: Sat, 04 Jun 2016 14:57:46 +0000
Subject: [Freeswitch-users] Dialplan Intercept/Eavesdrop Clarification
Message-ID: <CAPtU-UpSe2LA9BvCOhMGYSbQt+yRtFN2uY0QNkVCsdbBb9JLUg@mail.gmail.com>

Hey all,

I'm sure I'm missing something but I think I'm looking for a middle ground
between eavesdrop and intercept. Essentially I have a call running on
channel A. New channel B is created. I want channel B to take over all
media of channel A, and hangup channel A. Channel B should be put in the
exact spot that channel A was in at the time of the interception.

Theoretically, if I understand correctly, this is what intercept with -bleg
should do, but the call I'm intercepting may not be bridged to another leg.
It maybe in a conference, it may be running through the dialplan, etc. I
just want this new channel to continue in the same place as the current.

Is there a way I can get that behavior?

Thanks in advance.

Best,
Colin
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From bipin at xbipin.com  Sat Jun  4 19:23:18 2016
From: bipin at xbipin.com (Bipin Patel)
Date: Sat, 4 Jun 2016 19:23:18 +0400
Subject: [Freeswitch-users] mod_gsmopen wont compile on rpi3
Message-ID: <273d8483-9294-7679-3239-471b8f11ec30@xbipin.com>

sorry if this seems like a stupid question but im a newbie to linux, i 
compiled FS 1.6 on rpi3, later i wanted to add the mod_gsmopen module so 
i enabled that in modules and recompiled but it never got compiled so 
then i tried compiling it standalone using the below commands but make 
clean and make install keep failing with message no rule to make target 
clean/install

apt-get install gsm-utils
apt-get install libgsmme-dev
apt-get install usb-modeswitch-data usb-modeswitch
cd /usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/libctb-0.16/build
make DEBUG=0 GPIB=0
make DEBUG=0 GPIB=0 install
ldconfig
cd /usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/
make clean
make install



-- 
Regards,
Bipin


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



From krice at freeswitch.org  Sat Jun  4 20:06:23 2016
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 4 Jun 2016 11:06:23 -0500
Subject: [Freeswitch-users] mod_gsmopen wont compile on rpi3
In-Reply-To: <273d8483-9294-7679-3239-471b8f11ec30@xbipin.com>
References: <273d8483-9294-7679-3239-471b8f11ec30@xbipin.com>
Message-ID: <82099A62-32FF-4E71-9769-CC46C3ED2CF7@freeswitch.org>

Sounds like you need to open a jira and atach as a txt file the full build log so e the gsmopen dev can review

Sent from my iPhone

> On Jun 4, 2016, at 10:23 AM, Bipin Patel <bipin at xbipin.com> wrote:
> 
> sorry if this seems like a stupid question but im a newbie to linux, i 
> compiled FS 1.6 on rpi3, later i wanted to add the mod_gsmopen module so 
> i enabled that in modules and recompiled but it never got compiled so 
> then i tried compiling it standalone using the below commands but make 
> clean and make install keep failing with message no rule to make target 
> clean/install
> 
> apt-get install gsm-utils
> apt-get install libgsmme-dev
> apt-get install usb-modeswitch-data usb-modeswitch
> cd /usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/libctb-0.16/build
> make DEBUG=0 GPIB=0
> make DEBUG=0 GPIB=0 install
> ldconfig
> cd /usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/
> make clean
> make install
> 
> 
> 
> -- 
> Regards,
> Bipin
> 
> 
> ------------------------------------------------------------------------
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From Shawn.Wheeler at interlockconcepts.com  Sat Jun  4 20:11:57 2016
From: Shawn.Wheeler at interlockconcepts.com (Shawn Wheeler)
Date: Sat, 4 Jun 2016 16:11:57 +0000
Subject: [Freeswitch-users] Low cost voip phone ?
In-Reply-To: <CA+zebiYKUJTOsow4uRTH3keeKJh=TGpTQh=e5M529=nMLPX6oA@mail.gmail.com>
References: <xwblnhn6bbxbmii9dsid86wv.1464707474412@email.android.com>
	<CAHzfiZ7JkNMzWYGsm0A1vsyMmE0wLvj9rcRhPquCuPzfJKOTOA@mail.gmail.com>
	<5p4x48mh6rkjfx8v2f1smhrw.1464708465158@email.android.com>
	<CAH=YQ8skz9zzLnFzHLVMU9Cvt_GbgE3MWZRmj=zEQ6PMD4hXjA@mail.gmail.com>
	<jxhm3jmfynjdhvghugwcd774.1464709645365@email.android.com>
	<CAH=YQ8vTpW3kPBjzgedKCAYBpmRQ65PpOB-aYAwpZTvqdDB97g@mail.gmail.com>
	<p4acphb0apbwj94m3c8v992c.1464727543204@email.android.com>
	<CA+zebiYKUJTOsow4uRTH3keeKJh=TGpTQh=e5M529=nMLPX6oA@mail.gmail.com>
Message-ID: <SN1PR0201MB1504AC92F0D5CA5305037C31E05A0@SN1PR0201MB1504.namprd02.prod.outlook.com>

Jason,
This worked like a champ.  Thank you!


Shawn Wheeler
Chief Evangelist
shawn.wheeler at interlockconcepts.com<mailto:shawn.wheeler at interlockconcepts.com>
Cell:602.466.8855

[interlock concepts horizontal 600 x 205]<http://www.interlockconcecpts.com/>

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason Komar
Sent: Tuesday, May 31, 2016 1:50 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Low cost voip phone ?

If you are using only 4 numbers, try changing the dial plan in the phone to {xxxx}

Jason


On Tue, May 31, 2016 at 2:45 PM, Shawn Wheeler <Shawn.Wheeler at interlockconcepts.com<mailto:Shawn.Wheeler at interlockconcepts.com>> wrote:
It has the default setting in it at the moment.  And I am using 4 digits as this phone is for internal only.  I wl give send and or pound a try.  Thank you



Sent from my Verizon, Samsung Galaxy smartphone


-------- Original message --------
From: "happy.neko" <happy.neko at gmail.com<mailto:happy.neko at gmail.com>>
Date: 5/31/16 14:33 (GMT-07:00)
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] Low cost voip phone ?

2016-05-31 18:47 GMT+03:00 Shawn Wheeler <Shawn.Wheeler at interlockconcepts.com<mailto:Shawn.Wheeler at interlockconcepts.com>>:

I have heard about the dial plan and found it in the gui but I have not been successful in finding what it needs to be set to.

Dialplan sets how many digits device expects to be pressed before actually placing the call. You may configure it to set local numbers to be exactly 10 digits etc.
As Bob said try to press SEND or # key immediately after the phone number to see if it makes the difference.

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

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

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

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From gmaruzz at gmail.com  Sat Jun  4 20:52:09 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 4 Jun 2016 18:52:09 +0200
Subject: [Freeswitch-users] mod_gsmopen wont compile on rpi3
In-Reply-To: <82099A62-32FF-4E71-9769-CC46C3ED2CF7@freeswitch.org>
References: <273d8483-9294-7679-3239-471b8f11ec30@xbipin.com>
	<82099A62-32FF-4E71-9769-CC46C3ED2CF7@freeswitch.org>
Message-ID: <CALXCt0rS-nBNF+ENfPRS=XtN5NhY81LT1rg2bCCDZjAyYw_uzg@mail.gmail.com>

That is the old way to compile a module.

After all the steps you correctly took for libs, module enabling etc, try
this:

cd /usr/local/src/freeswitch
make install

it will probably works

-giovanni

On Sat, Jun 4, 2016 at 6:06 PM, Ken Rice <krice at freeswitch.org> wrote:

> Sounds like you need to open a jira and atach as a txt file the full build
> log so e the gsmopen dev can review
>
> Sent from my iPhone
>
> > On Jun 4, 2016, at 10:23 AM, Bipin Patel <bipin at xbipin.com> wrote:
> >
> > sorry if this seems like a stupid question but im a newbie to linux, i
> > compiled FS 1.6 on rpi3, later i wanted to add the mod_gsmopen module so
> > i enabled that in modules and recompiled but it never got compiled so
> > then i tried compiling it standalone using the below commands but make
> > clean and make install keep failing with message no rule to make target
> > clean/install
> >
> > apt-get install gsm-utils
> > apt-get install libgsmme-dev
> > apt-get install usb-modeswitch-data usb-modeswitch
> > cd
> /usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/libctb-0.16/build
> > make DEBUG=0 GPIB=0
> > make DEBUG=0 GPIB=0 install
> > ldconfig
> > cd /usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/
> > make clean
> > make install
> >
> >
> >
> > --
> > Regards,
> > Bipin
> >
> >
> > ------------------------------------------------------------------------
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From colin.morelli at gmail.com  Sat Jun  4 20:53:51 2016
From: colin.morelli at gmail.com (Colin Morelli)
Date: Sat, 04 Jun 2016 16:53:51 +0000
Subject: [Freeswitch-users] Bridge Call and Join Conference
In-Reply-To: <CAPtU-UpJMabAq_ngz6wpFScigO834o-ACw5QmgK1mLNrMcJJ_w@mail.gmail.com>
References: <CAPtU-UqO_jVO1OFxthu7TkqWCitUpwcU2E3SU0p0QmpXJ09MMQ@mail.gmail.com>
	<CAFmZFiMfBeH5XRa6KqFq27COr-h_YKWFKR4FQX6Vi-ej9WRPGQ@mail.gmail.com>
	<CAPtU-UpJMabAq_ngz6wpFScigO834o-ACw5QmgK1mLNrMcJJ_w@mail.gmail.com>
Message-ID: <CAPtU-Uo9Xsz4vW4xwqpvufc+mxT1G3=4nSXnegjdFFjH6g+Frg@mail.gmail.com>

Alright so after experimenting with this, it doesn't seem to work reliably
(though I may be using it wrong). Roughly half of the time everything works
fine and both parties are joined together. The other half of the time the
bridged leg doesn't make it to the conference (but also stays active). This
is the httapi response I'm providing:

<document type="xml/freeswitch-httapi">
   <work>
      <execute application="export" data="hangup_after_bridge=false" />
      <execute application="export" data="api_on_answer=uuid_transfer
${uuid} -both
'conference:2ee73476-aea4-4f85-861f-287f90a42275++flags{mintwo,dist-dtmf}'
inline" />
      <execute application="export" data="ringback=${us-ring}" />
      <execute application="bridge" data="sofia/external/
sips:number at sip.trunking.provider.com;transport=tls" />
   </work>
</document>

Would appreciate any insight.

Best,
Colin

On Wed, Jun 1, 2016 at 12:13 PM Colin Morelli <colin.morelli at gmail.com>
wrote:

> Awesome - this is exactly what I was looking for. Don't know how I missed
> it.
>
> Thank you!
>
> Best,
> Colin
>
> On Wed, Jun 1, 2016 at 12:08 PM Abaci B <abaci64 at gmail.com> wrote:
>
>> one way to do it would be to use api_on_answer
>> <https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-Theapi_onfamily>
>> to execute a uuid_transfer -both
>> <https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-uuid_transfer>
>> to transfer both legs to the conference (make sure to set
>> hangup_after_bridge to false).
>>
>> On Tue, May 31, 2016 at 4:59 PM, Colin Morelli <colin.morelli at gmail.com>
>> wrote:
>>
>>> So, I'm trying to setup certain calls such that they enter into a
>>> conference asap. I have the process working using bridging conferences
>>> right now, which is okay, but bridging conferences forces the dialing leg
>>> of the call to go to an answered state immediately, while the bridge leg is
>>> still ringing. So, while everything seems to work correctly, it messes with
>>> some of my reconciliation/reporting process with another database that are
>>> fed off of channel state events. Essentially it looks like every call was
>>> answered.
>>>
>>> So I was curious to know if there's a way to make a call that creates a
>>> bridged leg, and then as soon as the leg is answered immediately joins both
>>> legs into a conference. Is the only way to do it to listen for the
>>> channel_bridge events and then transfer the call myself?
>>>
>>> Best,
>>> Colin
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
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From colin.morelli at gmail.com  Sat Jun  4 21:00:20 2016
From: colin.morelli at gmail.com (Colin Morelli)
Date: Sat, 04 Jun 2016 17:00:20 +0000
Subject: [Freeswitch-users] Bridge Call and Join Conference
In-Reply-To: <CAPtU-Uo9Xsz4vW4xwqpvufc+mxT1G3=4nSXnegjdFFjH6g+Frg@mail.gmail.com>
References: <CAPtU-UqO_jVO1OFxthu7TkqWCitUpwcU2E3SU0p0QmpXJ09MMQ@mail.gmail.com>
	<CAFmZFiMfBeH5XRa6KqFq27COr-h_YKWFKR4FQX6Vi-ej9WRPGQ@mail.gmail.com>
	<CAPtU-UpJMabAq_ngz6wpFScigO834o-ACw5QmgK1mLNrMcJJ_w@mail.gmail.com>
	<CAPtU-Uo9Xsz4vW4xwqpvufc+mxT1G3=4nSXnegjdFFjH6g+Frg@mail.gmail.com>
Message-ID: <CAPtU-Ury1RYrPzdDPSVj0qQRuxiNB=SH5UeZis9Uscvgmez6Bg@mail.gmail.com>

Nevermind I think I figured it out. Using "export" instead of "set" caused
api_on_answer to be set on both channels. Combined with the {mintwo} flag
on the conference, what ended up happening was FS would execute the
transfer of both channels for the first channel, which would place both
callers into the conference. Then, it would execute the api_on_answer for
the bridged channel (because that variable was exported) which (depending
on when it ran) would pull the channel out of the conference and put it
back in, but pulling that user out would trip the mintwo flag and
apparently leave the call in a strange state.

Changing "export" to "set" seems to have solved the problem.

On Sat, Jun 4, 2016 at 12:53 PM Colin Morelli <colin.morelli at gmail.com>
wrote:

> Alright so after experimenting with this, it doesn't seem to work reliably
> (though I may be using it wrong). Roughly half of the time everything works
> fine and both parties are joined together. The other half of the time the
> bridged leg doesn't make it to the conference (but also stays active). This
> is the httapi response I'm providing:
>
> <document type="xml/freeswitch-httapi">
>    <work>
>       <execute application="export" data="hangup_after_bridge=false" />
>       <execute application="export" data="api_on_answer=uuid_transfer
> ${uuid} -both
> 'conference:2ee73476-aea4-4f85-861f-287f90a42275++flags{mintwo,dist-dtmf}'
> inline" />
>       <execute application="export" data="ringback=${us-ring}" />
>       <execute application="bridge" data="sofia/external/
> sips:number at sip.trunking.provider.com;transport=tls" />
>    </work>
> </document>
>
> Would appreciate any insight.
>
> Best,
> Colin
>
> On Wed, Jun 1, 2016 at 12:13 PM Colin Morelli <colin.morelli at gmail.com>
> wrote:
>
>> Awesome - this is exactly what I was looking for. Don't know how I missed
>> it.
>>
>> Thank you!
>>
>> Best,
>> Colin
>>
>> On Wed, Jun 1, 2016 at 12:08 PM Abaci B <abaci64 at gmail.com> wrote:
>>
>>> one way to do it would be to use api_on_answer
>>> <https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-Theapi_onfamily>
>>> to execute a uuid_transfer -both
>>> <https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-uuid_transfer>
>>> to transfer both legs to the conference (make sure to set
>>> hangup_after_bridge to false).
>>>
>>> On Tue, May 31, 2016 at 4:59 PM, Colin Morelli <colin.morelli at gmail.com>
>>> wrote:
>>>
>>>> So, I'm trying to setup certain calls such that they enter into a
>>>> conference asap. I have the process working using bridging conferences
>>>> right now, which is okay, but bridging conferences forces the dialing leg
>>>> of the call to go to an answered state immediately, while the bridge leg is
>>>> still ringing. So, while everything seems to work correctly, it messes with
>>>> some of my reconciliation/reporting process with another database that are
>>>> fed off of channel state events. Essentially it looks like every call was
>>>> answered.
>>>>
>>>> So I was curious to know if there's a way to make a call that creates a
>>>> bridged leg, and then as soon as the leg is answered immediately joins both
>>>> legs into a conference. Is the only way to do it to listen for the
>>>> channel_bridge events and then transfer the call myself?
>>>>
>>>> Best,
>>>> Colin
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
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From colin.morelli at gmail.com  Sat Jun  4 21:41:18 2016
From: colin.morelli at gmail.com (Colin Morelli)
Date: Sat, 04 Jun 2016 17:41:18 +0000
Subject: [Freeswitch-users] Bridge Call and Join Conference
In-Reply-To: <CAPtU-Ury1RYrPzdDPSVj0qQRuxiNB=SH5UeZis9Uscvgmez6Bg@mail.gmail.com>
References: <CAPtU-UqO_jVO1OFxthu7TkqWCitUpwcU2E3SU0p0QmpXJ09MMQ@mail.gmail.com>
	<CAFmZFiMfBeH5XRa6KqFq27COr-h_YKWFKR4FQX6Vi-ej9WRPGQ@mail.gmail.com>
	<CAPtU-UpJMabAq_ngz6wpFScigO834o-ACw5QmgK1mLNrMcJJ_w@mail.gmail.com>
	<CAPtU-Uo9Xsz4vW4xwqpvufc+mxT1G3=4nSXnegjdFFjH6g+Frg@mail.gmail.com>
	<CAPtU-Ury1RYrPzdDPSVj0qQRuxiNB=SH5UeZis9Uscvgmez6Bg@mail.gmail.com>
Message-ID: <CAPtU-UrThrVZwOp_SGoH5e=jRwmRYk0LVSeM1uAH+wh+Fq3m9w@mail.gmail.com>

Ugh and now I'm here to throw another wrench in this whole problem again.
Apparently this approach works *on some calls *and not on others. While I
can't yet identify why it works, it does consistently fail on some numbers.
For example, this works when dialing my cell phone (consistently). It fails
(consistently) when dialing the (804) 222-1111 test number

Looking through the call logs, it seems that in the case it works (with my
cell phone), uuid_transfer executes the transfer on the bridged channel *and
then* the originator channel. In the case where it doesn't work (the test
number), uuid_transfer executes the transfer on the originator channel, and
never touches the bridged channel. This results in the originate failing
with ORIGINATOR_CANCEL.

Is this a bug? It seems like for some reason the api_on_answer executes (on
some calls) possibly too early.

Best,
Colin

On Sat, Jun 4, 2016 at 1:00 PM Colin Morelli <colin.morelli at gmail.com>
wrote:

> Nevermind I think I figured it out. Using "export" instead of "set" caused
> api_on_answer to be set on both channels. Combined with the {mintwo} flag
> on the conference, what ended up happening was FS would execute the
> transfer of both channels for the first channel, which would place both
> callers into the conference. Then, it would execute the api_on_answer for
> the bridged channel (because that variable was exported) which (depending
> on when it ran) would pull the channel out of the conference and put it
> back in, but pulling that user out would trip the mintwo flag and
> apparently leave the call in a strange state.
>
> Changing "export" to "set" seems to have solved the problem.
>
> On Sat, Jun 4, 2016 at 12:53 PM Colin Morelli <colin.morelli at gmail.com>
> wrote:
>
>> Alright so after experimenting with this, it doesn't seem to work
>> reliably (though I may be using it wrong). Roughly half of the time
>> everything works fine and both parties are joined together. The other half
>> of the time the bridged leg doesn't make it to the conference (but also
>> stays active). This is the httapi response I'm providing:
>>
>> <document type="xml/freeswitch-httapi">
>>    <work>
>>       <execute application="export" data="hangup_after_bridge=false" />
>>       <execute application="export" data="api_on_answer=uuid_transfer
>> ${uuid} -both
>> 'conference:2ee73476-aea4-4f85-861f-287f90a42275++flags{mintwo,dist-dtmf}'
>> inline" />
>>       <execute application="export" data="ringback=${us-ring}" />
>>       <execute application="bridge" data="sofia/external/
>> sips:number at sip.trunking.provider.com;transport=tls" />
>>    </work>
>> </document>
>>
>> Would appreciate any insight.
>>
>> Best,
>> Colin
>>
>> On Wed, Jun 1, 2016 at 12:13 PM Colin Morelli <colin.morelli at gmail.com>
>> wrote:
>>
>>> Awesome - this is exactly what I was looking for. Don't know how I
>>> missed it.
>>>
>>> Thank you!
>>>
>>> Best,
>>> Colin
>>>
>>> On Wed, Jun 1, 2016 at 12:08 PM Abaci B <abaci64 at gmail.com> wrote:
>>>
>>>> one way to do it would be to use api_on_answer
>>>> <https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-Theapi_onfamily>
>>>> to execute a uuid_transfer -both
>>>> <https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-uuid_transfer>
>>>> to transfer both legs to the conference (make sure to set
>>>> hangup_after_bridge to false).
>>>>
>>>> On Tue, May 31, 2016 at 4:59 PM, Colin Morelli <colin.morelli at gmail.com
>>>> > wrote:
>>>>
>>>>> So, I'm trying to setup certain calls such that they enter into a
>>>>> conference asap. I have the process working using bridging conferences
>>>>> right now, which is okay, but bridging conferences forces the dialing leg
>>>>> of the call to go to an answered state immediately, while the bridge leg is
>>>>> still ringing. So, while everything seems to work correctly, it messes with
>>>>> some of my reconciliation/reporting process with another database that are
>>>>> fed off of channel state events. Essentially it looks like every call was
>>>>> answered.
>>>>>
>>>>> So I was curious to know if there's a way to make a call that creates
>>>>> a bridged leg, and then as soon as the leg is answered immediately joins
>>>>> both legs into a conference. Is the only way to do it to listen for the
>>>>> channel_bridge events and then transfer the call myself?
>>>>>
>>>>> Best,
>>>>> Colin
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
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From colin.morelli at gmail.com  Sat Jun  4 22:17:33 2016
From: colin.morelli at gmail.com (Colin Morelli)
Date: Sat, 04 Jun 2016 18:17:33 +0000
Subject: [Freeswitch-users] Bridge Call and Join Conference
In-Reply-To: <CAPtU-UrThrVZwOp_SGoH5e=jRwmRYk0LVSeM1uAH+wh+Fq3m9w@mail.gmail.com>
References: <CAPtU-UqO_jVO1OFxthu7TkqWCitUpwcU2E3SU0p0QmpXJ09MMQ@mail.gmail.com>
	<CAFmZFiMfBeH5XRa6KqFq27COr-h_YKWFKR4FQX6Vi-ej9WRPGQ@mail.gmail.com>
	<CAPtU-UpJMabAq_ngz6wpFScigO834o-ACw5QmgK1mLNrMcJJ_w@mail.gmail.com>
	<CAPtU-Uo9Xsz4vW4xwqpvufc+mxT1G3=4nSXnegjdFFjH6g+Frg@mail.gmail.com>
	<CAPtU-Ury1RYrPzdDPSVj0qQRuxiNB=SH5UeZis9Uscvgmez6Bg@mail.gmail.com>
	<CAPtU-UrThrVZwOp_SGoH5e=jRwmRYk0LVSeM1uAH+wh+Fq3m9w@mail.gmail.com>
Message-ID: <CAPtU-UqAM91ni6j-yDm5xTZvtbj5ycMN=2Mu3Dif0twQrbWfvw@mail.gmail.com>

Sorry for blowing everyone up with emails - but hopefully this will be
helpful to someone else in the future.

As it turns out, api_on_answer is entirely disconnected from the bridging
concept (makes sense now that I look into how this all works more). It was
working when calling my cell phone because I was receiving early media
(ringing) from the SIP trunking provider. This resulted in the bridge
starting so that, by the time api_on_answer executed, there was a bridged
leg. In the case of the test number, however, I receive no early media. As
a result, when the channel was answered, the bridge hadn't started yet, so
there was no remote leg to transfer.

After digging through the source, I discovered, "api_before_bridge" which
appears to execute *after* the channels have have their peer sessions
setup, as part of the bridge loop, whether early media was present or not.
This now seems to be working consistently, as I expect it to.

I'd still really appreciate someone letting me know if this is good/bad, or
if you know of a better approach.

Best,
Colin

On Sat, Jun 4, 2016 at 1:41 PM Colin Morelli <colin.morelli at gmail.com>
wrote:

> Ugh and now I'm here to throw another wrench in this whole problem again.
> Apparently this approach works *on some calls *and not on others. While I
> can't yet identify why it works, it does consistently fail on some numbers.
> For example, this works when dialing my cell phone (consistently). It fails
> (consistently) when dialing the (804) 222-1111 test number
>
> Looking through the call logs, it seems that in the case it works (with my
> cell phone), uuid_transfer executes the transfer on the bridged channel *and
> then* the originator channel. In the case where it doesn't work (the test
> number), uuid_transfer executes the transfer on the originator channel, and
> never touches the bridged channel. This results in the originate failing
> with ORIGINATOR_CANCEL.
>
> Is this a bug? It seems like for some reason the api_on_answer executes
> (on some calls) possibly too early.
>
> Best,
> Colin
>
> On Sat, Jun 4, 2016 at 1:00 PM Colin Morelli <colin.morelli at gmail.com>
> wrote:
>
>> Nevermind I think I figured it out. Using "export" instead of "set"
>> caused api_on_answer to be set on both channels. Combined with the {mintwo}
>> flag on the conference, what ended up happening was FS would execute the
>> transfer of both channels for the first channel, which would place both
>> callers into the conference. Then, it would execute the api_on_answer for
>> the bridged channel (because that variable was exported) which (depending
>> on when it ran) would pull the channel out of the conference and put it
>> back in, but pulling that user out would trip the mintwo flag and
>> apparently leave the call in a strange state.
>>
>> Changing "export" to "set" seems to have solved the problem.
>>
>> On Sat, Jun 4, 2016 at 12:53 PM Colin Morelli <colin.morelli at gmail.com>
>> wrote:
>>
>>> Alright so after experimenting with this, it doesn't seem to work
>>> reliably (though I may be using it wrong). Roughly half of the time
>>> everything works fine and both parties are joined together. The other half
>>> of the time the bridged leg doesn't make it to the conference (but also
>>> stays active). This is the httapi response I'm providing:
>>>
>>> <document type="xml/freeswitch-httapi">
>>>    <work>
>>>       <execute application="export" data="hangup_after_bridge=false" />
>>>       <execute application="export" data="api_on_answer=uuid_transfer
>>> ${uuid} -both
>>> 'conference:2ee73476-aea4-4f85-861f-287f90a42275++flags{mintwo,dist-dtmf}'
>>> inline" />
>>>       <execute application="export" data="ringback=${us-ring}" />
>>>       <execute application="bridge" data="sofia/external/
>>> sips:number at sip.trunking.provider.com;transport=tls" />
>>>    </work>
>>> </document>
>>>
>>> Would appreciate any insight.
>>>
>>> Best,
>>> Colin
>>>
>>> On Wed, Jun 1, 2016 at 12:13 PM Colin Morelli <colin.morelli at gmail.com>
>>> wrote:
>>>
>>>> Awesome - this is exactly what I was looking for. Don't know how I
>>>> missed it.
>>>>
>>>> Thank you!
>>>>
>>>> Best,
>>>> Colin
>>>>
>>>> On Wed, Jun 1, 2016 at 12:08 PM Abaci B <abaci64 at gmail.com> wrote:
>>>>
>>>>> one way to do it would be to use api_on_answer
>>>>> <https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-Theapi_onfamily>
>>>>> to execute a uuid_transfer -both
>>>>> <https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-uuid_transfer>
>>>>> to transfer both legs to the conference (make sure to set
>>>>> hangup_after_bridge to false).
>>>>>
>>>>> On Tue, May 31, 2016 at 4:59 PM, Colin Morelli <
>>>>> colin.morelli at gmail.com> wrote:
>>>>>
>>>>>> So, I'm trying to setup certain calls such that they enter into a
>>>>>> conference asap. I have the process working using bridging conferences
>>>>>> right now, which is okay, but bridging conferences forces the dialing leg
>>>>>> of the call to go to an answered state immediately, while the bridge leg is
>>>>>> still ringing. So, while everything seems to work correctly, it messes with
>>>>>> some of my reconciliation/reporting process with another database that are
>>>>>> fed off of channel state events. Essentially it looks like every call was
>>>>>> answered.
>>>>>>
>>>>>> So I was curious to know if there's a way to make a call that creates
>>>>>> a bridged leg, and then as soon as the leg is answered immediately joins
>>>>>> both legs into a conference. Is the only way to do it to listen for the
>>>>>> channel_bridge events and then transfer the call myself?
>>>>>>
>>>>>> Best,
>>>>>> Colin
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
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From ssinyagin at gmail.com  Sat Jun  4 22:53:43 2016
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sat, 4 Jun 2016 20:53:43 +0200
Subject: [Freeswitch-users] SwitchPi
In-Reply-To: <C3945830-AF48-4982-8B66-ABE04C6D0CB6@freeswitch.org>
References: <CABV9BUu8ygpbbGLzxmzG70NsiS7dj5D4WRkNaQH6Hzex_Q38yg@mail.gmail.com>
	<00bb01d1bc20$ab399cb0$01acd610$@freeswitch.org>
	<CABV9BUuns9Z+-KYJvj8LeRrDUkKBnL-Dh46dXJ-Y2kDFent2Rg@mail.gmail.com>
	<04ea01d1bcf1$c1a373f0$44ea5bd0$@freeswitch.org>
	<CAMDdSzMj5-GU1M1930ueDFGMDiZPTwUqgVFr=Gj2ukGxT2-W2w@mail.gmail.com>
	<095c01d1bda2$2159f430$640ddc90$@freeswitch.org>
	<CACOYK=gucwrvYqJRLCP8x9w6EQg2L8RwqG8SZV2y=u+bYLjPiA@mail.gmail.com>
	<C3945830-AF48-4982-8B66-ABE04C6D0CB6@freeswitch.org>
Message-ID: <CACOYK=jKPU4UtTwFsuixTzxbLgmK6-9+1PEWd3_LyiGnpNfwdg@mail.gmail.com>

It's not about endianness, but pointer alignment.
On Jun 4, 2016 02:54, "Ken Rice" <krice at freeswitch.org> wrote:

> I asked steve about this and yes i have duplicated it. Looks like possibly
> an endian issue
>
> Sent from my iPhone
>
> On Jun 3, 2016, at 7:11 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
>
> Ken, you probably hit this bug with faxing:
> https://freeswitch.org/jira/browse/FS-8651
>
> we're still waiting for Steve's feedback on that one.
>
>
>
>
>
> On Fri, Jun 3, 2016 at 4:13 PM, Ken Rice <krice at freeswitch.org> wrote:
>
>> Why the Pi you ask? Why custom hardware to go with it?
>>
>>
>>
>> Generic X86 Hardware is not the end all be all. While the X86 based CPUs
>> means that there can be less portability issues than say something like ARM
>> or SPARC or other architectures, the PI still typically offers several
>> other capabilities that may not be offered on the other Single Board
>> Computers at a $35 price point.
>>
>>
>>
>> Then there is the well documented pile of GPIO ports that make it quite
>> easy to drive things like custom keypads, servos, steppers, etc etc etc?
>>  this means you can spin a small custom board (in my case its about 1.5
>> square inches) that has the specific additional hardware for adding an
>> additional 24 GPIOs giving room for a combination of something like 36
>> buttons worth of keypads and 12 other dry contact inputs and outputs (if
>> you use 12 of the GPIO pins to drive a keypad array you can get 36 buttons.
>> Basic formula  for max number of potential keys   (n/2)^2 where n is the
>> number of GPIO pins used in the keypad array.
>>
>>
>>
>> Now mix that with that fact that ARM is not a second class Linux citizen
>> anymore, we can pretty easily copy the source over compile and go. And
>> things just work (barring the occasional endian issue that happens when
>> taking software between architectures that are big vs little vs bi endian)
>>
>>
>>
>> Now theres always talk of a new cheap x86 based SoC such as the Edison,
>> or other Arm based SBC like the odriod, or the CHIP, but I have yet to see
>> one that is as readily available as the Pi at the $35 with the myriad of
>> addons and software already running on it.
>>
>> (Yesterday I installed FreeSWITCH on a Pi and it took about 45 minutes?
>> there was 1 issues with Faxing that we are looking into and its probably an
>> endian issue)
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Colton
>> Conor
>> *Sent:* Friday, June 3, 2016 7:59 AM
>>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* Re: [Freeswitch-users] SwitchPi
>>
>>
>>
>> May I ask the question of why make something customer just to run on the
>> Rasberry Pi? There are X86 SOC coming to the market that will be close to
>> the cost of the Pi, and be much more powerful. Plus, you can run regular
>> x86 operating systems.
>>
>>
>>
>> On Thu, Jun 2, 2016 at 12:11 PM, Ken Rice <krice at freeswitch.org> wrote:
>>
>> The custom boards allow for hardware DTMF keypad interface, hook switch
>> and a few other things like LCD for CLID on old school phones with that
>> provision.
>>
>>
>>
>> There will be official Raspian packages sooner or later? theres a know
>> issue with the cowbuilder that?s blocking this at this time, we?re working
>> on a work around?
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *William
>> Scott
>> *Sent:* Wednesday, June 1, 2016 6:52 PM
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* Re: [Freeswitch-users] SwitchPi
>>
>>
>>
>> Hi
>>
>>
>>
>> Yes, I have FusionPBX/freeSwitch running on a RPi3. Will the ARM build
>> ever become "Official"?
>>
>>
>>
>> I've watched a 'tube video or two. What is the functionality of the
>> custom boards?
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bipin at xbipin.com  Sat Jun  4 23:53:02 2016
From: bipin at xbipin.com (Bipin Patel)
Date: Sat, 04 Jun 2016 23:53:02 +0400
Subject: [Freeswitch-users] mod_gsmopen wont compile on rpi3
In-Reply-To: <CALXCt0rS-nBNF+ENfPRS=XtN5NhY81LT1rg2bCCDZjAyYw_uzg@mail.gmail.com>
References: <273d8483-9294-7679-3239-471b8f11ec30@xbipin.com>
	<82099A62-32FF-4E71-9769-CC46C3ED2CF7@freeswitch.org>
	<CALXCt0rS-nBNF+ENfPRS=XtN5NhY81LT1rg2bCCDZjAyYw_uzg@mail.gmail.com>
Message-ID: <1551cf9d130.2787.b07ebdf329620b8089087c7205b03f01@xbipin.com>

Thanks for the reply. Well after enabling in modules I had run the rpi3 fs 
install script which didn't compile it so I ran those old commands but that 
didn't help so let me try this method. I enable in modules file and 
directly run that command or do I need to run any other commands like 
clean, configure or any thing else?




On June 4, 2016 8:53:58 PM Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:

> That is the old way to compile a module.
>
> After all the steps you correctly took for libs, module enabling etc, try
> this:
>
> cd /usr/local/src/freeswitch
> make install
>
> it will probably works
>
> -giovanni
>
> On Sat, Jun 4, 2016 at 6:06 PM, Ken Rice <krice at freeswitch.org> wrote:
>
>> Sounds like you need to open a jira and atach as a txt file the full build
>> log so e the gsmopen dev can review
>>
>> Sent from my iPhone
>>
>> > On Jun 4, 2016, at 10:23 AM, Bipin Patel <bipin at xbipin.com> wrote:
>> >
>> > sorry if this seems like a stupid question but im a newbie to linux, i
>> > compiled FS 1.6 on rpi3, later i wanted to add the mod_gsmopen module so
>> > i enabled that in modules and recompiled but it never got compiled so
>> > then i tried compiling it standalone using the below commands but make
>> > clean and make install keep failing with message no rule to make target
>> > clean/install
>> >
>> > apt-get install gsm-utils
>> > apt-get install libgsmme-dev
>> > apt-get install usb-modeswitch-data usb-modeswitch
>> > cd
>> /usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/libctb-0.16/build
>> > make DEBUG=0 GPIB=0
>> > make DEBUG=0 GPIB=0 install
>> > ldconfig
>> > cd /usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/
>> > make clean
>> > make install
>> >
>> >
>> >
>> > --
>> > Regards,
>> > Bipin
>> >
>> >
>> > ------------------------------------------------------------------------
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
>
>
> ----------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From gmaruzz at gmail.com  Sun Jun  5 00:25:27 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 4 Jun 2016 22:25:27 +0200
Subject: [Freeswitch-users] mod_gsmopen wont compile on rpi3
In-Reply-To: <1551cf9d130.2787.b07ebdf329620b8089087c7205b03f01@xbipin.com>
References: <273d8483-9294-7679-3239-471b8f11ec30@xbipin.com>
	<82099A62-32FF-4E71-9769-CC46C3ED2CF7@freeswitch.org>
	<CALXCt0rS-nBNF+ENfPRS=XtN5NhY81LT1rg2bCCDZjAyYw_uzg@mail.gmail.com>
	<1551cf9d130.2787.b07ebdf329620b8089087c7205b03f01@xbipin.com>
Message-ID: <CALXCt0r+3vo2Wisq2yg_JQXrUcNaJdeKC9dUqSeD2=8Sj_UY_w@mail.gmail.com>

After making libs, and modules.conf,  just make install

sent from mobile
cell: +39 347 266 56 18
Giovanni Maruzzelli
OpenTelecom.IT
Il 04/Giu/2016 21:53, "Bipin Patel" <bipin at xbipin.com> ha scritto:

> Thanks for the reply. Well after enabling in modules I had run the rpi3 fs
> install script which didn't compile it so I ran those old commands but that
> didn't help so let me try this method. I enable in modules file and
> directly run that command or do I need to run any other commands like
> clean, configure or any thing else?
>
> On June 4, 2016 8:53:58 PM Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
>
>> That is the old way to compile a module.
>>
>> After all the steps you correctly took for libs, module enabling etc, try
>> this:
>>
>> cd /usr/local/src/freeswitch
>> make install
>>
>> it will probably works
>>
>> -giovanni
>>
>> On Sat, Jun 4, 2016 at 6:06 PM, Ken Rice <krice at freeswitch.org> wrote:
>>
>>> Sounds like you need to open a jira and atach as a txt file the full
>>> build log so e the gsmopen dev can review
>>>
>>> Sent from my iPhone
>>>
>>> > On Jun 4, 2016, at 10:23 AM, Bipin Patel <bipin at xbipin.com> wrote:
>>> >
>>> > sorry if this seems like a stupid question but im a newbie to linux, i
>>> > compiled FS 1.6 on rpi3, later i wanted to add the mod_gsmopen module
>>> so
>>> > i enabled that in modules and recompiled but it never got compiled so
>>> > then i tried compiling it standalone using the below commands but make
>>> > clean and make install keep failing with message no rule to make target
>>> > clean/install
>>> >
>>> > apt-get install gsm-utils
>>> > apt-get install libgsmme-dev
>>> > apt-get install usb-modeswitch-data usb-modeswitch
>>> > cd
>>> /usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/libctb-0.16/build
>>> > make DEBUG=0 GPIB=0
>>> > make DEBUG=0 GPIB=0 install
>>> > ldconfig
>>> > cd /usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/
>>> > make clean
>>> > make install
>>> >
>>> >
>>> >
>>> > --
>>> > Regards,
>>> > Bipin
>>> >
>>> >
>>> >
>>> ------------------------------------------------------------------------
>>> >
>>> >
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://confluence.freeswitch.org
>>> > http://www.cluecon.com
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From william at williamcollsassoc.ca  Sun Jun  5 01:25:57 2016
From: william at williamcollsassoc.ca (William Colls)
Date: Sat, 4 Jun 2016 17:25:57 -0400
Subject: [Freeswitch-users] Domain and Context
Message-ID: <57534765.7070403@williamcollsassoc.ca>


This is really a very basic question, but it is confusing me.

If I want to have my users in a particular domain, do I need to create 
the domain entry in the directory, and then the various contexts within 
the domain, or the other way around?

e.g /usr/local/freeswitch/conf/directory/domain/context

  or

/usr/local/freeswitch/conf/directory/context/domain

Thanks for your time.

William.


From ba.lerest at gmail.com  Sat Jun  4 16:48:08 2016
From: ba.lerest at gmail.com (Bastien LE REST)
Date: Sat, 4 Jun 2016 13:48:08 +0100
Subject: [Freeswitch-users] ICE issue
Message-ID: <CAL+i1s9-BPaWvmXnsEAmVV24kd_tB-Mz8a8NW7n=mQXvR0gZMQ@mail.gmail.com>

Hi,

When I enable ICE using PJSIP, FreeSwitch answers "488 Not Acceptable
Here". When I disable ICE, I don't have any problems, the calls work just
fine. I tried to use another SIP client with ICE and the call works fine.
So I think the problem has to do with PJSIP.

I'm working on the migration from an Asterisk architecture to a FreeSwitch
architecture. I have to deal with PJSIP and ICE support because my client
is a mobile application who is already published on the stores.

PJSIP trace: https://pastebin.freeswitch.org/view/ddeaae4b
FreeSwitch trace: https://pastebin.freeswitch.org/view/d380e01d

Could you please have a look at the trace and tell me what am I doing
wrong? Thank you in advance.

Kind regards,
Bastien.
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From bipin at xbipin.com  Sun Jun  5 10:07:58 2016
From: bipin at xbipin.com (Bipin Patel)
Date: Sun, 5 Jun 2016 10:07:58 +0400
Subject: [Freeswitch-users] mod_gsmopen wont compile on rpi3
In-Reply-To: <CALXCt0r+3vo2Wisq2yg_JQXrUcNaJdeKC9dUqSeD2=8Sj_UY_w@mail.gmail.com>
References: <273d8483-9294-7679-3239-471b8f11ec30@xbipin.com>
	<82099A62-32FF-4E71-9769-CC46C3ED2CF7@freeswitch.org>
	<CALXCt0rS-nBNF+ENfPRS=XtN5NhY81LT1rg2bCCDZjAyYw_uzg@mail.gmail.com>
	<1551cf9d130.2787.b07ebdf329620b8089087c7205b03f01@xbipin.com>
	<CALXCt0r+3vo2Wisq2yg_JQXrUcNaJdeKC9dUqSeD2=8Sj_UY_w@mail.gmail.com>
Message-ID: <0584270f-b90a-c0f2-da84-b55a472e6fde@xbipin.com>

i still get this

pi at raspberrypi:/usr/local/src/freeswitch $ make install
make: *** No rule to make target 'install'.  Stop.
pi at raspberrypi:/usr/local/src/freeswitch $ sudo make install
make: *** No rule to make target 'install'.  Stop.



Regards,
Bipin


------------------------------------------------------------------------
-------- Original Message --------
Subject: Re: [Freeswitch-users] mod_gsmopen wont compile on rpi3
From: Giovanni Maruzzelli <gmaruzz at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date: 6/5/2016, 12:25:27 AM
>
> After making libs, and modules.conf,  just make install
>
> sent from mobile
> cell: +39 347 266 56 18
> Giovanni Maruzzelli
> OpenTelecom.IT
>
> Il 04/Giu/2016 21:53, "Bipin Patel" <bipin at xbipin.com 
> <mailto:bipin at xbipin.com>> ha scritto:
>
>     Thanks for the reply. Well after enabling in modules I had run the
>     rpi3 fs install script which didn't compile it so I ran those old
>     commands but that didn't help so let me try this method. I enable
>     in modules file and directly run that command or do I need to run
>     any other commands like clean, configure or any thing else?
>
>     On June 4, 2016 8:53:58 PM Giovanni Maruzzelli <gmaruzz at gmail.com
>     <mailto:gmaruzz at gmail.com>> wrote:
>
>>     That is the old way to compile a module.
>>
>>     After all the steps you correctly took for libs, module enabling
>>     etc, try this:
>>
>>     cd /usr/local/src/freeswitch
>>     make install
>>
>>     it will probably works
>>
>>     -giovanni
>>
>>     On Sat, Jun 4, 2016 at 6:06 PM, Ken Rice <krice at freeswitch.org
>>     <mailto:krice at freeswitch.org>> wrote:
>>
>>         Sounds like you need to open a jira and atach as a txt file
>>         the full build log so e the gsmopen dev can review
>>
>>         Sent from my iPhone
>>
>>         > On Jun 4, 2016, at 10:23 AM, Bipin Patel <bipin at xbipin.com
>>         <mailto:bipin at xbipin.com>> wrote:
>>         >
>>         > sorry if this seems like a stupid question but im a newbie
>>         to linux, i
>>         > compiled FS 1.6 on rpi3, later i wanted to add the
>>         mod_gsmopen module so
>>         > i enabled that in modules and recompiled but it never got
>>         compiled so
>>         > then i tried compiling it standalone using the below
>>         commands but make
>>         > clean and make install keep failing with message no rule to
>>         make target
>>         > clean/install
>>         >
>>         > apt-get install gsm-utils
>>         > apt-get install libgsmme-dev
>>         > apt-get install usb-modeswitch-data usb-modeswitch
>>         > cd
>>         /usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/libctb-0.16/build
>>         > make DEBUG=0 GPIB=0
>>         > make DEBUG=0 GPIB=0 install
>>         > ldconfig
>>         > cd /usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/
>>         > make clean
>>         > make install
>>         >
>>         >
>>         >
>>         > --
>>         > Regards,
>>         > Bipin
>>         >
>>         >
>>         >
>>         ------------------------------------------------------------------------
>>         >
>>         >
>>         >
>>         _________________________________________________________________________
>>         > Professional FreeSWITCH Consulting Services:
>>         > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>         > http://www.freeswitchsolutions.com
>>         >
>>         > Official FreeSWITCH Sites
>>         > http://www.freeswitch.org
>>         > http://confluence.freeswitch.org
>>         > http://www.cluecon.com
>>         >
>>         > FreeSWITCH-users mailing list
>>         > FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         >
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         > http://www.freeswitch.org
>>
>>         _________________________________________________________________________
>>         Professional FreeSWITCH Consulting Services:
>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>         http://www.freeswitchsolutions.com
>>
>>         Official FreeSWITCH Sites
>>         http://www.freeswitch.org
>>         http://confluence.freeswitch.org
>>         http://www.cluecon.com
>>
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>
>>     -- 
>>     Sincerely,
>>
>>     Giovanni Maruzzelli
>>     Cell : +39-347-2665618 <tel:%2B39-347-2665618>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org <mailto:consulting%40freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://confluence.freeswitch.org
>>     http://www.cluecon.com
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users%40lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From lexxua at gmail.com  Sun Jun  5 12:16:53 2016
From: lexxua at gmail.com (Volodymyr Fedorov)
Date: Sun, 5 Jun 2016 10:16:53 +0200
Subject: [Freeswitch-users] ICE issue
In-Reply-To: <CAL+i1s9-BPaWvmXnsEAmVV24kd_tB-Mz8a8NW7n=mQXvR0gZMQ@mail.gmail.com>
References: <CAL+i1s9-BPaWvmXnsEAmVV24kd_tB-Mz8a8NW7n=mQXvR0gZMQ@mail.gmail.com>
Message-ID: <CAO_GGAp3pJmk=ixtgyDgg0C0eDVB2OrVd2jcztZ=oKiM1mTGGA@mail.gmail.com>

Hi, seems you did not enable STUN server in your PJSIP-client config and
you sending only lan ip addresses as ICE-candidates.

Br,
Volodymyr

On Sat, Jun 4, 2016 at 2:48 PM, Bastien LE REST <ba.lerest at gmail.com> wrote:

> Hi,
>
> When I enable ICE using PJSIP, FreeSwitch answers "488 Not Acceptable
> Here". When I disable ICE, I don't have any problems, the calls work just
> fine. I tried to use another SIP client with ICE and the call works fine.
> So I think the problem has to do with PJSIP.
>
> I'm working on the migration from an Asterisk architecture to a FreeSwitch
> architecture. I have to deal with PJSIP and ICE support because my client
> is a mobile application who is already published on the stores.
>
> PJSIP trace: https://pastebin.freeswitch.org/view/ddeaae4b
> FreeSwitch trace: https://pastebin.freeswitch.org/view/d380e01d
>
> Could you please have a look at the trace and tell me what am I doing
> wrong? Thank you in advance.
>
> Kind regards,
> Bastien.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Best regards,
Volodymyr
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From gmaruzz at gmail.com  Sun Jun  5 16:02:00 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sun, 5 Jun 2016 14:02:00 +0200
Subject: [Freeswitch-users] mod_gsmopen wont compile on rpi3
In-Reply-To: <0584270f-b90a-c0f2-da84-b55a472e6fde@xbipin.com>
References: <273d8483-9294-7679-3239-471b8f11ec30@xbipin.com>
	<82099A62-32FF-4E71-9769-CC46C3ED2CF7@freeswitch.org>
	<CALXCt0rS-nBNF+ENfPRS=XtN5NhY81LT1rg2bCCDZjAyYw_uzg@mail.gmail.com>
	<1551cf9d130.2787.b07ebdf329620b8089087c7205b03f01@xbipin.com>
	<CALXCt0r+3vo2Wisq2yg_JQXrUcNaJdeKC9dUqSeD2=8Sj_UY_w@mail.gmail.com>
	<0584270f-b90a-c0f2-da84-b55a472e6fde@xbipin.com>
Message-ID: <CALXCt0rMbv_fobLYrkLPzVioscP9T+rE_KbQED5DFiv=TP6Mrg@mail.gmail.com>

I do not know how it works on raspberry, in a normal (x86) build you can do
make install

If you are using a special script to install on raspberry, please contact
the script developer.

-giovanni

On Sun, Jun 5, 2016 at 8:07 AM, Bipin Patel <bipin at xbipin.com> wrote:

> i still get this
>
> pi at raspberrypi:/usr/local/src/freeswitch $ make install
> make: *** No rule to make target 'install'.  Stop.
> pi at raspberrypi:/usr/local/src/freeswitch $ sudo make install
> make: *** No rule to make target 'install'.  Stop.
>
>
>
> Regards,
> Bipin
>
>
> ------------------------------------------------------------------------
> -------- Original Message --------
> Subject: Re: [Freeswitch-users] mod_gsmopen wont compile on rpi3
> From: Giovanni Maruzzelli <gmaruzz at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: 6/5/2016, 12:25:27 AM
> >
> > After making libs, and modules.conf,  just make install
> >
> > sent from mobile
> > cell: +39 347 266 56 18
> > Giovanni Maruzzelli
> > OpenTelecom.IT
> >
> > Il 04/Giu/2016 21:53, "Bipin Patel" <bipin at xbipin.com
> > <mailto:bipin at xbipin.com>> ha scritto:
> >
> >     Thanks for the reply. Well after enabling in modules I had run the
> >     rpi3 fs install script which didn't compile it so I ran those old
> >     commands but that didn't help so let me try this method. I enable
> >     in modules file and directly run that command or do I need to run
> >     any other commands like clean, configure or any thing else?
> >
> >     On June 4, 2016 8:53:58 PM Giovanni Maruzzelli <gmaruzz at gmail.com
> >     <mailto:gmaruzz at gmail.com>> wrote:
> >
> >>     That is the old way to compile a module.
> >>
> >>     After all the steps you correctly took for libs, module enabling
> >>     etc, try this:
> >>
> >>     cd /usr/local/src/freeswitch
> >>     make install
> >>
> >>     it will probably works
> >>
> >>     -giovanni
> >>
> >>     On Sat, Jun 4, 2016 at 6:06 PM, Ken Rice <krice at freeswitch.org
> >>     <mailto:krice at freeswitch.org>> wrote:
> >>
> >>         Sounds like you need to open a jira and atach as a txt file
> >>         the full build log so e the gsmopen dev can review
> >>
> >>         Sent from my iPhone
> >>
> >>         > On Jun 4, 2016, at 10:23 AM, Bipin Patel <bipin at xbipin.com
> >>         <mailto:bipin at xbipin.com>> wrote:
> >>         >
> >>         > sorry if this seems like a stupid question but im a newbie
> >>         to linux, i
> >>         > compiled FS 1.6 on rpi3, later i wanted to add the
> >>         mod_gsmopen module so
> >>         > i enabled that in modules and recompiled but it never got
> >>         compiled so
> >>         > then i tried compiling it standalone using the below
> >>         commands but make
> >>         > clean and make install keep failing with message no rule to
> >>         make target
> >>         > clean/install
> >>         >
> >>         > apt-get install gsm-utils
> >>         > apt-get install libgsmme-dev
> >>         > apt-get install usb-modeswitch-data usb-modeswitch
> >>         > cd
> >>
>  /usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/libctb-0.16/build
> >>         > make DEBUG=0 GPIB=0
> >>         > make DEBUG=0 GPIB=0 install
> >>         > ldconfig
> >>         > cd /usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/
> >>         > make clean
> >>         > make install
> >>         >
> >>         >
> >>         >
> >>         > --
> >>         > Regards,
> >>         > Bipin
> >>         >
> >>         >
> >>         >
> >>
>  ------------------------------------------------------------------------
> >>         >
> >>         >
> >>         >
> >>
>  _________________________________________________________________________
> >>         > Professional FreeSWITCH Consulting Services:
> >>         > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >>         > http://www.freeswitchsolutions.com
> >>         >
> >>         > Official FreeSWITCH Sites
> >>         > http://www.freeswitch.org
> >>         > http://confluence.freeswitch.org
> >>         > http://www.cluecon.com
> >>         >
> >>         > FreeSWITCH-users mailing list
> >>         > FreeSWITCH-users at lists.freeswitch.org
> >>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >>         > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>         >
> >>         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>         > http://www.freeswitch.org
> >>
> >>
>  _________________________________________________________________________
> >>         Professional FreeSWITCH Consulting Services:
> >>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >>         http://www.freeswitchsolutions.com
> >>
> >>         Official FreeSWITCH Sites
> >>         http://www.freeswitch.org
> >>         http://confluence.freeswitch.org
> >>         http://www.cluecon.com
> >>
> >>         FreeSWITCH-users mailing list
> >>         FreeSWITCH-users at lists.freeswitch.org
> >>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>         http://www.freeswitch.org
> >>
> >>
> >>
> >>
> >>     --
> >>     Sincerely,
> >>
> >>     Giovanni Maruzzelli
> >>     Cell : +39-347-2665618 <tel:%2B39-347-2665618>
> >>
>  _________________________________________________________________________
> >>     Professional FreeSWITCH Consulting Services:
> >>     consulting at freeswitch.org <mailto:consulting%40freeswitch.org>
> >>     http://www.freeswitchsolutions.com
> >>
> >>     Official FreeSWITCH Sites
> >>     http://www.freeswitch.org
> >>     http://confluence.freeswitch.org
> >>     http://www.cluecon.com
> >>
> >>     FreeSWITCH-users mailing list
> >>     FreeSWITCH-users at lists.freeswitch.org
> >>     <mailto:FreeSWITCH-users%40lists.freeswitch.org>
> >>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>     http://www.freeswitch.org
> >
> >
>  _________________________________________________________________________
> >     Professional FreeSWITCH Consulting Services:
> >     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >     http://www.freeswitchsolutions.com
> >
> >     Official FreeSWITCH Sites
> >     http://www.freeswitch.org
> >     http://confluence.freeswitch.org
> >     http://www.cluecon.com
> >
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From bipin at xbipin.com  Sun Jun  5 16:11:12 2016
From: bipin at xbipin.com (Bipin Patel)
Date: Sun, 5 Jun 2016 16:11:12 +0400
Subject: [Freeswitch-users] mod_gsmopen wont compile on rpi3
In-Reply-To: <CALXCt0rMbv_fobLYrkLPzVioscP9T+rE_KbQED5DFiv=TP6Mrg@mail.gmail.com>
References: <273d8483-9294-7679-3239-471b8f11ec30@xbipin.com>
	<82099A62-32FF-4E71-9769-CC46C3ED2CF7@freeswitch.org>
	<CALXCt0rS-nBNF+ENfPRS=XtN5NhY81LT1rg2bCCDZjAyYw_uzg@mail.gmail.com>
	<1551cf9d130.2787.b07ebdf329620b8089087c7205b03f01@xbipin.com>
	<CALXCt0r+3vo2Wisq2yg_JQXrUcNaJdeKC9dUqSeD2=8Sj_UY_w@mail.gmail.com>
	<0584270f-b90a-c0f2-da84-b55a472e6fde@xbipin.com>
	<CALXCt0rMbv_fobLYrkLPzVioscP9T+rE_KbQED5DFiv=TP6Mrg@mail.gmail.com>
Message-ID: <dd6cf3d9-d53c-32b1-5712-1abbfc152530@xbipin.com>

its the script which FS staff had created but anyways i figured it all 
out and it got compiled and working now, thanks



Regards,
Bipin


------------------------------------------------------------------------
-------- Original Message --------
Subject: Re: [Freeswitch-users] mod_gsmopen wont compile on rpi3
From: Giovanni Maruzzelli <gmaruzz at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date: 6/5/2016, 4:02:00 PM
> I do not know how it works on raspberry, in a normal (x86) build you 
> can do make install
>
> If you are using a special script to install on raspberry, please 
> contact the script developer.
>
> -giovanni
>
> On Sun, Jun 5, 2016 at 8:07 AM, Bipin Patel <bipin at xbipin.com 
> <mailto:bipin at xbipin.com>> wrote:
>
>     i still get this
>
>     pi at raspberrypi:/usr/local/src/freeswitch $ make install
>     make: *** No rule to make target 'install'.  Stop.
>     pi at raspberrypi:/usr/local/src/freeswitch $ sudo make install
>     make: *** No rule to make target 'install'.  Stop.
>
>
>
>     Regards,
>     Bipin
>
>
>     ------------------------------------------------------------------------
>     -------- Original Message --------
>     Subject: Re: [Freeswitch-users] mod_gsmopen wont compile on rpi3
>     From: Giovanni Maruzzelli <gmaruzz at gmail.com
>     <mailto:gmaruzz at gmail.com>>
>     To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>
>     Date: 6/5/2016, 12:25:27 AM
>     >
>     > After making libs, and modules.conf,  just make install
>     >
>     > sent from mobile
>     > cell: +39 347 266 56 18 <tel:%2B39%20347%20266%2056%2018>
>     > Giovanni Maruzzelli
>     > OpenTelecom.IT
>     >
>     > Il 04/Giu/2016 21:53, "Bipin Patel" <bipin at xbipin.com
>     <mailto:bipin at xbipin.com>
>     > <mailto:bipin at xbipin.com <mailto:bipin at xbipin.com>>> ha scritto:
>     >
>     >     Thanks for the reply. Well after enabling in modules I had
>     run the
>     >     rpi3 fs install script which didn't compile it so I ran
>     those old
>     >     commands but that didn't help so let me try this method. I
>     enable
>     >     in modules file and directly run that command or do I need
>     to run
>     >     any other commands like clean, configure or any thing else?
>     >
>     >     On June 4, 2016 8:53:58 PM Giovanni Maruzzelli
>     <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>
>     >     <mailto:gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>>> wrote:
>     >
>     >>     That is the old way to compile a module.
>     >>
>     >>     After all the steps you correctly took for libs, module
>     enabling
>     >>     etc, try this:
>     >>
>     >>     cd /usr/local/src/freeswitch
>     >>     make install
>     >>
>     >>     it will probably works
>     >>
>     >>     -giovanni
>     >>
>     >>     On Sat, Jun 4, 2016 at 6:06 PM, Ken Rice
>     <krice at freeswitch.org <mailto:krice at freeswitch.org>
>     >>     <mailto:krice at freeswitch.org <mailto:krice at freeswitch.org>>> wrote:
>     >>
>     >>         Sounds like you need to open a jira and atach as a txt file
>     >>         the full build log so e the gsmopen dev can review
>     >>
>     >>         Sent from my iPhone
>     >>
>     >>         > On Jun 4, 2016, at 10:23 AM, Bipin Patel
>     <bipin at xbipin.com <mailto:bipin at xbipin.com>
>     >>         <mailto:bipin at xbipin.com <mailto:bipin at xbipin.com>>> wrote:
>     >>         >
>     >>         > sorry if this seems like a stupid question but im a
>     newbie
>     >>         to linux, i
>     >>         > compiled FS 1.6 on rpi3, later i wanted to add the
>     >>         mod_gsmopen module so
>     >>         > i enabled that in modules and recompiled but it never got
>     >>         compiled so
>     >>         > then i tried compiling it standalone using the below
>     >>         commands but make
>     >>         > clean and make install keep failing with message no
>     rule to
>     >>         make target
>     >>         > clean/install
>     >>         >
>     >>         > apt-get install gsm-utils
>     >>         > apt-get install libgsmme-dev
>     >>         > apt-get install usb-modeswitch-data usb-modeswitch
>     >>         > cd
>     >>
>      /usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/libctb-0.16/build
>     >>         > make DEBUG=0 GPIB=0
>     >>         > make DEBUG=0 GPIB=0 install
>     >>         > ldconfig
>     >>         > cd
>     /usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/
>     >>         > make clean
>     >>         > make install
>     >>         >
>     >>         >
>     >>         >
>     >>         > --
>     >>         > Regards,
>     >>         > Bipin
>     >>         >
>     >>         >
>     >>         >
>     >>
>      ------------------------------------------------------------------------
>     >>         >
>     >>         >
>     >>         >
>     >>
>      _________________________________________________________________________
>     >>         > Professional FreeSWITCH Consulting Services:
>     >>         > consulting at freeswitch.org
>     <mailto:consulting at freeswitch.org>
>     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>
>     >>         >http://www.freeswitchsolutions.com
>     >>         >
>     >>         > Official FreeSWITCH Sites
>     >>         > http://www.freeswitch.org
>     >>         > http://confluence.freeswitch.org
>     >>         > http://www.cluecon.com
>     >>         >
>     >>         > FreeSWITCH-users mailing list
>     >>         > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >>         <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >>         >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>         >
>     >>       
>      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >>         > http://www.freeswitch.org
>     >>
>     >>
>      _________________________________________________________________________
>     >>         Professional FreeSWITCH Consulting Services:
>     >> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>
>     >>http://www.freeswitchsolutions.com
>     >>
>     >>         Official FreeSWITCH Sites
>     >> http://www.freeswitch.org
>     >> http://confluence.freeswitch.org
>     >> http://www.cluecon.com
>     >>
>     >>         FreeSWITCH-users mailing list
>     >> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >>         <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>       
>      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >> http://www.freeswitch.org
>     >>
>     >>
>     >>
>     >>
>     >>     --
>     >>     Sincerely,
>     >>
>     >>     Giovanni Maruzzelli
>     >>     Cell : +39-347-2665618 <tel:%2B39-347-2665618>
>     <tel:%2B39-347-2665618>
>     >> _________________________________________________________________________
>     >>     Professional FreeSWITCH Consulting Services:
>     >> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     <mailto:consulting%40freeswitch.org
>     <mailto:consulting%2540freeswitch.org>>
>     >>http://www.freeswitchsolutions.com
>     >>
>     >>     Official FreeSWITCH Sites
>     >> http://www.freeswitch.org
>     >> http://confluence.freeswitch.org
>     >> http://www.cluecon.com
>     >>
>     >>     FreeSWITCH-users mailing list
>     >> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >>     <mailto:FreeSWITCH-users%40lists.freeswitch.org
>     <mailto:FreeSWITCH-users%2540lists.freeswitch.org>>
>     >>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>   
>      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >> http://www.freeswitch.org
>     >
>     >
>      _________________________________________________________________________
>     >     Professional FreeSWITCH Consulting Services:
>     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>
>     >http://www.freeswitchsolutions.com
>     >
>     >     Official FreeSWITCH Sites
>     > http://www.freeswitch.org
>     > http://confluence.freeswitch.org
>     > http://www.cluecon.com
>     >
>     >     FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >   
>      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>     >
>     >
>     >
>     >
>     _________________________________________________________________________
>     > Professional FreeSWITCH Consulting Services:
>     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     > http://www.freeswitchsolutions.com
>     >
>     > Official FreeSWITCH Sites
>     > http://www.freeswitch.org
>     > http://confluence.freeswitch.org
>     > http://www.cluecon.com
>     >
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From gmaruzz at gmail.com  Sun Jun  5 16:40:17 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sun, 5 Jun 2016 14:40:17 +0200
Subject: [Freeswitch-users] mod_gsmopen wont compile on rpi3
In-Reply-To: <dd6cf3d9-d53c-32b1-5712-1abbfc152530@xbipin.com>
References: <273d8483-9294-7679-3239-471b8f11ec30@xbipin.com>
	<82099A62-32FF-4E71-9769-CC46C3ED2CF7@freeswitch.org>
	<CALXCt0rS-nBNF+ENfPRS=XtN5NhY81LT1rg2bCCDZjAyYw_uzg@mail.gmail.com>
	<1551cf9d130.2787.b07ebdf329620b8089087c7205b03f01@xbipin.com>
	<CALXCt0r+3vo2Wisq2yg_JQXrUcNaJdeKC9dUqSeD2=8Sj_UY_w@mail.gmail.com>
	<0584270f-b90a-c0f2-da84-b55a472e6fde@xbipin.com>
	<CALXCt0rMbv_fobLYrkLPzVioscP9T+rE_KbQED5DFiv=TP6Mrg@mail.gmail.com>
	<dd6cf3d9-d53c-32b1-5712-1abbfc152530@xbipin.com>
Message-ID: <CALXCt0oyJ+QeQwQU=66NLPaSn6bEXSfA_dMrhhH-6jJkr6ccVw@mail.gmail.com>

can you please document it and detail it for other people who can have same
problem?

Please, describe all steps for installing it.

Thanks in advance
-giovanni

On Sun, Jun 5, 2016 at 2:11 PM, Bipin Patel <bipin at xbipin.com> wrote:

> its the script which FS staff had created but anyways i figured it all
> out and it got compiled and working now, thanks
>
>
>
> Regards,
> Bipin
>
>
> ------------------------------------------------------------------------
> -------- Original Message --------
> Subject: Re: [Freeswitch-users] mod_gsmopen wont compile on rpi3
> From: Giovanni Maruzzelli <gmaruzz at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: 6/5/2016, 4:02:00 PM
> > I do not know how it works on raspberry, in a normal (x86) build you
> > can do make install
> >
> > If you are using a special script to install on raspberry, please
> > contact the script developer.
> >
> > -giovanni
> >
> > On Sun, Jun 5, 2016 at 8:07 AM, Bipin Patel <bipin at xbipin.com
> > <mailto:bipin at xbipin.com>> wrote:
> >
> >     i still get this
> >
> >     pi at raspberrypi:/usr/local/src/freeswitch $ make install
> >     make: *** No rule to make target 'install'.  Stop.
> >     pi at raspberrypi:/usr/local/src/freeswitch $ sudo make install
> >     make: *** No rule to make target 'install'.  Stop.
> >
> >
> >
> >     Regards,
> >     Bipin
> >
> >
> >
>  ------------------------------------------------------------------------
> >     -------- Original Message --------
> >     Subject: Re: [Freeswitch-users] mod_gsmopen wont compile on rpi3
> >     From: Giovanni Maruzzelli <gmaruzz at gmail.com
> >     <mailto:gmaruzz at gmail.com>>
> >     To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org
> >     <mailto:freeswitch-users at lists.freeswitch.org>>
> >     Date: 6/5/2016, 12:25:27 AM
> >     >
> >     > After making libs, and modules.conf,  just make install
> >     >
> >     > sent from mobile
> >     > cell: +39 347 266 56 18 <tel:%2B39%20347%20266%2056%2018>
> >     > Giovanni Maruzzelli
> >     > OpenTelecom.IT
> >     >
> >     > Il 04/Giu/2016 21:53, "Bipin Patel" <bipin at xbipin.com
> >     <mailto:bipin at xbipin.com>
> >     > <mailto:bipin at xbipin.com <mailto:bipin at xbipin.com>>> ha scritto:
> >     >
> >     >     Thanks for the reply. Well after enabling in modules I had
> >     run the
> >     >     rpi3 fs install script which didn't compile it so I ran
> >     those old
> >     >     commands but that didn't help so let me try this method. I
> >     enable
> >     >     in modules file and directly run that command or do I need
> >     to run
> >     >     any other commands like clean, configure or any thing else?
> >     >
> >     >     On June 4, 2016 8:53:58 PM Giovanni Maruzzelli
> >     <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>
> >     >     <mailto:gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>>> wrote:
> >     >
> >     >>     That is the old way to compile a module.
> >     >>
> >     >>     After all the steps you correctly took for libs, module
> >     enabling
> >     >>     etc, try this:
> >     >>
> >     >>     cd /usr/local/src/freeswitch
> >     >>     make install
> >     >>
> >     >>     it will probably works
> >     >>
> >     >>     -giovanni
> >     >>
> >     >>     On Sat, Jun 4, 2016 at 6:06 PM, Ken Rice
> >     <krice at freeswitch.org <mailto:krice at freeswitch.org>
> >     >>     <mailto:krice at freeswitch.org <mailto:krice at freeswitch.org>>>
> wrote:
> >     >>
> >     >>         Sounds like you need to open a jira and atach as a txt
> file
> >     >>         the full build log so e the gsmopen dev can review
> >     >>
> >     >>         Sent from my iPhone
> >     >>
> >     >>         > On Jun 4, 2016, at 10:23 AM, Bipin Patel
> >     <bipin at xbipin.com <mailto:bipin at xbipin.com>
> >     >>         <mailto:bipin at xbipin.com <mailto:bipin at xbipin.com>>>
> wrote:
> >     >>         >
> >     >>         > sorry if this seems like a stupid question but im a
> >     newbie
> >     >>         to linux, i
> >     >>         > compiled FS 1.6 on rpi3, later i wanted to add the
> >     >>         mod_gsmopen module so
> >     >>         > i enabled that in modules and recompiled but it never
> got
> >     >>         compiled so
> >     >>         > then i tried compiling it standalone using the below
> >     >>         commands but make
> >     >>         > clean and make install keep failing with message no
> >     rule to
> >     >>         make target
> >     >>         > clean/install
> >     >>         >
> >     >>         > apt-get install gsm-utils
> >     >>         > apt-get install libgsmme-dev
> >     >>         > apt-get install usb-modeswitch-data usb-modeswitch
> >     >>         > cd
> >     >>
> >
> /usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/libctb-0.16/build
> >     >>         > make DEBUG=0 GPIB=0
> >     >>         > make DEBUG=0 GPIB=0 install
> >     >>         > ldconfig
> >     >>         > cd
> >     /usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/
> >     >>         > make clean
> >     >>         > make install
> >     >>         >
> >     >>         >
> >     >>         >
> >     >>         > --
> >     >>         > Regards,
> >     >>         > Bipin
> >     >>         >
> >     >>         >
> >     >>         >
> >     >>
> >
> ------------------------------------------------------------------------
> >     >>         >
> >     >>         >
> >     >>         >
> >     >>
> >
> _________________________________________________________________________
> >     >>         > Professional FreeSWITCH Consulting Services:
> >     >>         > consulting at freeswitch.org
> >     <mailto:consulting at freeswitch.org>
> >     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org
> >>
> >     >>         >http://www.freeswitchsolutions.com
> >     >>         >
> >     >>         > Official FreeSWITCH Sites
> >     >>         > http://www.freeswitch.org
> >     >>         > http://confluence.freeswitch.org
> >     >>         > http://www.cluecon.com
> >     >>         >
> >     >>         > FreeSWITCH-users mailing list
> >     >>         > FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     >>         <mailto:FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
> >     >>         >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >>         >
> >     >>
> >      UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     >>         > http://www.freeswitch.org
> >     >>
> >     >>
> >
> _________________________________________________________________________
> >     >>         Professional FreeSWITCH Consulting Services:
> >     >> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org
> >>
> >     >>http://www.freeswitchsolutions.com
> >     >>
> >     >>         Official FreeSWITCH Sites
> >     >> http://www.freeswitch.org
> >     >> http://confluence.freeswitch.org
> >     >> http://www.cluecon.com
> >     >>
> >     >>         FreeSWITCH-users mailing list
> >     >> FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     >>         <mailto:FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
> >     >>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >>
> >      UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     >> http://www.freeswitch.org
> >     >>
> >     >>
> >     >>
> >     >>
> >     >>     --
> >     >>     Sincerely,
> >     >>
> >     >>     Giovanni Maruzzelli
> >     >>     Cell : +39-347-2665618 <tel:%2B39-347-2665618>
> >     <tel:%2B39-347-2665618>
> >     >>
> _________________________________________________________________________
> >     >>     Professional FreeSWITCH Consulting Services:
> >     >> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >     <mailto:consulting%40freeswitch.org
> >     <mailto:consulting%2540freeswitch.org>>
> >     >>http://www.freeswitchsolutions.com
> >     >>
> >     >>     Official FreeSWITCH Sites
> >     >> http://www.freeswitch.org
> >     >> http://confluence.freeswitch.org
> >     >> http://www.cluecon.com
> >     >>
> >     >>     FreeSWITCH-users mailing list
> >     >> FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     >>     <mailto:FreeSWITCH-users%40lists.freeswitch.org
> >     <mailto:FreeSWITCH-users%2540lists.freeswitch.org>>
> >     >>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >>
> >      UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     >> http://www.freeswitch.org
> >     >
> >     >
> >
> _________________________________________________________________________
> >     >     Professional FreeSWITCH Consulting Services:
> >     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org
> >>
> >     >http://www.freeswitchsolutions.com
> >     >
> >     >     Official FreeSWITCH Sites
> >     > http://www.freeswitch.org
> >     > http://confluence.freeswitch.org
> >     > http://www.cluecon.com
> >     >
> >     >     FreeSWITCH-users mailing list
> >     > FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
> >     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >
> >      UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     > http://www.freeswitch.org
> >     >
> >     >
> >     >
> >     >
> >
>  _________________________________________________________________________
> >     > Professional FreeSWITCH Consulting Services:
> >     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >     > http://www.freeswitchsolutions.com
> >     >
> >     > Official FreeSWITCH Sites
> >     > http://www.freeswitch.org
> >     > http://confluence.freeswitch.org
> >     > http://www.cluecon.com
> >     >
> >     > FreeSWITCH-users mailing list
> >     > FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     > http://www.freeswitch.org
> >
> >
> >
> >
>  _________________________________________________________________________
> >     Professional FreeSWITCH Consulting Services:
> >     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >     http://www.freeswitchsolutions.com
> >
> >     Official FreeSWITCH Sites
> >     http://www.freeswitch.org
> >     http://confluence.freeswitch.org
> >     http://www.cluecon.com
> >
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Sincerely,
> >
> > Giovanni Maruzzelli
> > Cell : +39-347-2665618
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From ssinyagin at gmail.com  Sun Jun  5 16:45:45 2016
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sun, 5 Jun 2016 14:45:45 +0200
Subject: [Freeswitch-users] Domain and Context
In-Reply-To: <57534765.7070403@williamcollsassoc.ca>
References: <57534765.7070403@williamcollsassoc.ca>
Message-ID: <CACOYK=iR+0L22YojZvwVV_OuB8=qCcC9Pee=9hHin+ty9vKxVQ@mail.gmail.com>

See here, I tried to explain it as shortly as possible:

https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/7144332
On Jun 4, 2016 23:26, "William Colls" <william at williamcollsassoc.ca> wrote:

>
> This is really a very basic question, but it is confusing me.
>
> If I want to have my users in a particular domain, do I need to create
> the domain entry in the directory, and then the various contexts within
> the domain, or the other way around?
>
> e.g /usr/local/freeswitch/conf/directory/domain/context
>
>   or
>
> /usr/local/freeswitch/conf/directory/context/domain
>
> Thanks for your time.
>
> William.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ssinyagin at gmail.com  Sun Jun  5 18:05:48 2016
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sun, 5 Jun 2016 16:05:48 +0200
Subject: [Freeswitch-users] mod_gsmopen: are USB dongles the only option?
Message-ID: <CACOYK=g=DrJipsvG6jg8yqUjC+bd5rhsUfgRH7eOf+tsF14=fQ@mail.gmail.com>

does anyone know a GSM/UMTS/LTE modem that has mini-PCIe interface and
is able to deliver voice signal that can be passed to mod_gsmopen?

The only one that has any kind of voice signal that I found, is Huawei
MU609 and the like. But they deliver PCM voice as a raw signal on
mini-PCIe pins which are marked as "reserved" in the interface
specifications, so normally the boards don't have any wiring for those
pins.

I just don't like the idea of an USB dongle, and there are many
hardware platforms with mini-PCIe interfaces. They work perfectly for
mobile data, but I couldn't find any option for voice.


From ba.lerest at gmail.com  Sun Jun  5 15:10:36 2016
From: ba.lerest at gmail.com (Bastien LE REST)
Date: Sun, 5 Jun 2016 12:10:36 +0100
Subject: [Freeswitch-users] ICE issue
In-Reply-To: <CAO_GGAp3pJmk=ixtgyDgg0C0eDVB2OrVd2jcztZ=oKiM1mTGGA@mail.gmail.com>
References: <CAL+i1s9-BPaWvmXnsEAmVV24kd_tB-Mz8a8NW7n=mQXvR0gZMQ@mail.gmail.com>
	<CAO_GGAp3pJmk=ixtgyDgg0C0eDVB2OrVd2jcztZ=oKiM1mTGGA@mail.gmail.com>
Message-ID: <CAL+i1s-4wWWnsZ+g+ZnrLdoe=KPAcD7j4Yq6bpNa28c=Ut1uig@mail.gmail.com>

Hi,

Thank you for your answer. You are true, when I enable a STUN server in my
PJSIP the calls work fine. I don't understand how the calls can work with
this configuration through Asterisk and not through FreeSwitch.

Kind regards,
Bastien.

On 5 June 2016 at 09:16, Volodymyr Fedorov <lexxua at gmail.com> wrote:

> Hi, seems you did not enable STUN server in your PJSIP-client config and
> you sending only lan ip addresses as ICE-candidates.
>
> Br,
> Volodymyr
>
> On Sat, Jun 4, 2016 at 2:48 PM, Bastien LE REST <ba.lerest at gmail.com>
> wrote:
>
>> Hi,
>>
>> When I enable ICE using PJSIP, FreeSwitch answers "488 Not Acceptable
>> Here". When I disable ICE, I don't have any problems, the calls work just
>> fine. I tried to use another SIP client with ICE and the call works fine.
>> So I think the problem has to do with PJSIP.
>>
>> I'm working on the migration from an Asterisk architecture to a
>> FreeSwitch architecture. I have to deal with PJSIP and ICE support because
>> my client is a mobile application who is already published on the stores.
>>
>> PJSIP trace: https://pastebin.freeswitch.org/view/ddeaae4b
>> FreeSwitch trace: https://pastebin.freeswitch.org/view/d380e01d
>>
>> Could you please have a look at the trace and tell me what am I doing
>> wrong? Thank you in advance.
>>
>> Kind regards,
>> Bastien.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Best regards,
> Volodymyr
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bipin at xbipin.com  Sun Jun  5 18:31:18 2016
From: bipin at xbipin.com (Bipin Patel)
Date: Sun, 5 Jun 2016 18:31:18 +0400
Subject: [Freeswitch-users] mod_gsmopen wont compile on rpi3
In-Reply-To: <CALXCt0oyJ+QeQwQU=66NLPaSn6bEXSfA_dMrhhH-6jJkr6ccVw@mail.gmail.com>
References: <273d8483-9294-7679-3239-471b8f11ec30@xbipin.com>
	<82099A62-32FF-4E71-9769-CC46C3ED2CF7@freeswitch.org>
	<CALXCt0rS-nBNF+ENfPRS=XtN5NhY81LT1rg2bCCDZjAyYw_uzg@mail.gmail.com>
	<1551cf9d130.2787.b07ebdf329620b8089087c7205b03f01@xbipin.com>
	<CALXCt0r+3vo2Wisq2yg_JQXrUcNaJdeKC9dUqSeD2=8Sj_UY_w@mail.gmail.com>
	<0584270f-b90a-c0f2-da84-b55a472e6fde@xbipin.com>
	<CALXCt0rMbv_fobLYrkLPzVioscP9T+rE_KbQED5DFiv=TP6Mrg@mail.gmail.com>
	<dd6cf3d9-d53c-32b1-5712-1abbfc152530@xbipin.com>
	<CALXCt0oyJ+QeQwQU=66NLPaSn6bEXSfA_dMrhhH-6jJkr6ccVw@mail.gmail.com>
Message-ID: <0997b238-18b4-c653-8eba-87d75d7cec05@xbipin.com>

actually i tried a lot of things so dont remember what caused it but i 
think these steps made it work, most important is to make the libctb 
first then run ldconfig as root then compile the whole of FS, these 
steps assume u have built FS with the defaults first which means 
./bootstrap.sh -j was already run once.

giovanni, can u confirm if all these commands r in the correct order, if 
so then ill add it to the gsmopen page because being a linux newbie i 
dont even know what a make and configure commands does after all and 
still have no clue if its even possible to build FS for the first time 
with the mod_gsmopen enabled in modules, that's y the steps assuming FS 
was built with the defaults first :)


first go edit the modules.conf file and uncomment the mod_gsmopen

sudo apt-get install gsm-utils
sudo apt-get install libgsmme-dev
sudo apt-get install usb-modeswitch-data usb-modeswitch

cd /usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/libctb-0.16/build
sudo make DEBUG=0 GPIB=0
sudo make DEBUG=0 GPIB=0 install
sudo ldconfig

cd /usr/local/src/freeswitch/
sudo ./configure -C
sudo make -j3
sudo make install


Regards,
Bipin


------------------------------------------------------------------------
-------- Original Message --------
Subject: Re: [Freeswitch-users] mod_gsmopen wont compile on rpi3
From: Giovanni Maruzzelli <gmaruzz at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date: 6/5/2016, 4:40:17 PM
> can you please document it and detail it for other people who can have 
> same problem?
>
> Please, describe all steps for installing it.
>
> Thanks in advance
> -giovanni
>
> On Sun, Jun 5, 2016 at 2:11 PM, Bipin Patel <bipin at xbipin.com 
> <mailto:bipin at xbipin.com>> wrote:
>
>     its the script which FS staff had created but anyways i figured it all
>     out and it got compiled and working now, thanks
>
>
>
>     Regards,
>     Bipin
>
>
>     ------------------------------------------------------------------------
>     -------- Original Message --------
>     Subject: Re: [Freeswitch-users] mod_gsmopen wont compile on rpi3
>     From: Giovanni Maruzzelli <gmaruzz at gmail.com
>     <mailto:gmaruzz at gmail.com>>
>     To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>
>     Date: 6/5/2016, 4:02:00 PM
>     > I do not know how it works on raspberry, in a normal (x86) build you
>     > can do make install
>     >
>     > If you are using a special script to install on raspberry, please
>     > contact the script developer.
>     >
>     > -giovanni
>     >
>     > On Sun, Jun 5, 2016 at 8:07 AM, Bipin Patel <bipin at xbipin.com
>     <mailto:bipin at xbipin.com>
>     > <mailto:bipin at xbipin.com <mailto:bipin at xbipin.com>>> wrote:
>     >
>     >     i still get this
>     >
>     >     pi at raspberrypi:/usr/local/src/freeswitch $ make install
>     >     make: *** No rule to make target 'install'. Stop.
>     >     pi at raspberrypi:/usr/local/src/freeswitch $ sudo make install
>     >     make: *** No rule to make target 'install'. Stop.
>     >
>     >
>     >
>     >     Regards,
>     >     Bipin
>     >
>     >
>     >
>      ------------------------------------------------------------------------
>     >     -------- Original Message --------
>     >     Subject: Re: [Freeswitch-users] mod_gsmopen wont compile on rpi3
>     >     From: Giovanni Maruzzelli <gmaruzz at gmail.com
>     <mailto:gmaruzz at gmail.com>
>     >     <mailto:gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>>>
>     >     To: FreeSWITCH Users Help
>     <freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     >     <mailto:freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>>
>     >     Date: 6/5/2016, 12:25:27 AM
>     >     >
>     >     > After making libs, and modules.conf,  just make install
>     >     >
>     >     > sent from mobile
>     >     > cell: +39 347 266 56 18 <tel:%2B39%20347%20266%2056%2018>
>     <tel:%2B39%20347%20266%2056%2018>
>     >     > Giovanni Maruzzelli
>     >     > OpenTelecom.IT
>     >     >
>     >     > Il 04/Giu/2016 21:53, "Bipin Patel" <bipin at xbipin.com
>     <mailto:bipin at xbipin.com>
>     >     <mailto:bipin at xbipin.com <mailto:bipin at xbipin.com>>
>     >     > <mailto:bipin at xbipin.com <mailto:bipin at xbipin.com>
>     <mailto:bipin at xbipin.com <mailto:bipin at xbipin.com>>>> ha scritto:
>     >     >
>     >     >     Thanks for the reply. Well after enabling in modules I had
>     >     run the
>     >     >     rpi3 fs install script which didn't compile it so I ran
>     >     those old
>     >     >     commands but that didn't help so let me try this method. I
>     >     enable
>     >     >     in modules file and directly run that command or do I need
>     >     to run
>     >     >     any other commands like clean, configure or any thing
>     else?
>     >     >
>     >     >     On June 4, 2016 8:53:58 PM Giovanni Maruzzelli
>     >     <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>
>     <mailto:gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>>
>     >     >     <mailto:gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>
>     <mailto:gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>>>> wrote:
>     >     >
>     >     >>     That is the old way to compile a module.
>     >     >>
>     >     >>     After all the steps you correctly took for libs, module
>     >     enabling
>     >     >>     etc, try this:
>     >     >>
>     >     >>     cd /usr/local/src/freeswitch
>     >     >>     make install
>     >     >>
>     >     >>     it will probably works
>     >     >>
>     >     >>     -giovanni
>     >     >>
>     >     >>     On Sat, Jun 4, 2016 at 6:06 PM, Ken Rice
>     >     <krice at freeswitch.org <mailto:krice at freeswitch.org>
>     <mailto:krice at freeswitch.org <mailto:krice at freeswitch.org>>
>     >     >>     <mailto:krice at freeswitch.org <mailto:krice at freeswitch.org>
>     <mailto:krice at freeswitch.org <mailto:krice at freeswitch.org>>>> wrote:
>     >     >>
>     >     >>         Sounds like you need to open a jira and atach as
>     a txt file
>     >     >>         the full build log so e the gsmopen dev can review
>     >     >>
>     >     >>         Sent from my iPhone
>     >     >>
>     >     >>         > On Jun 4, 2016, at 10:23 AM, Bipin Patel
>     >     <bipin at xbipin.com <mailto:bipin at xbipin.com>
>     <mailto:bipin at xbipin.com <mailto:bipin at xbipin.com>>
>     >     >>         <mailto:bipin at xbipin.com
>     <mailto:bipin at xbipin.com> <mailto:bipin at xbipin.com
>     <mailto:bipin at xbipin.com>>>> wrote:
>     >     >>         >
>     >     >>         > sorry if this seems like a stupid question but im a
>     >     newbie
>     >     >>         to linux, i
>     >     >>         > compiled FS 1.6 on rpi3, later i wanted to add the
>     >     >>         mod_gsmopen module so
>     >     >>         > i enabled that in modules and recompiled but it
>     never got
>     >     >>         compiled so
>     >     >>         > then i tried compiling it standalone using the
>     below
>     >     >>         commands but make
>     >     >>         > clean and make install keep failing with message no
>     >     rule to
>     >     >>         make target
>     >     >>         > clean/install
>     >     >>         >
>     >     >>         > apt-get install gsm-utils
>     >     >>         > apt-get install libgsmme-dev
>     >     >>         > apt-get install usb-modeswitch-data usb-modeswitch
>     >     >>         > cd
>     >     >>
>     >
>     /usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/libctb-0.16/build
>     >     >>         > make DEBUG=0 GPIB=0
>     >     >>         > make DEBUG=0 GPIB=0 install
>     >     >>         > ldconfig
>     >     >>         > cd
>     >  /usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/
>     >     >>         > make clean
>     >     >>         > make install
>     >     >>         >
>     >     >>         >
>     >     >>         >
>     >     >>         > --
>     >     >>         > Regards,
>     >     >>         > Bipin
>     >     >>         >
>     >     >>         >
>     >     >>         >
>     >     >>
>     >
>     ------------------------------------------------------------------------
>     >     >>         >
>     >     >>         >
>     >     >>         >
>     >     >>
>     >
>     _________________________________________________________________________
>     >     >>         > Professional FreeSWITCH Consulting Services:
>     >     >>         > consulting at freeswitch.org
>     <mailto:consulting at freeswitch.org>
>     >     <mailto:consulting at freeswitch.org
>     <mailto:consulting at freeswitch.org>>
>     >     <mailto:consulting at freeswitch.org
>     <mailto:consulting at freeswitch.org>
>     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>>
>     >     >>         >http://www.freeswitchsolutions.com
>     >     >>         >
>     >     >>         > Official FreeSWITCH Sites
>     >     >>         > http://www.freeswitch.org
>     >     >>         > http://confluence.freeswitch.org
>     >     >>         > http://www.cluecon.com
>     >     >>         >
>     >     >>         > FreeSWITCH-users mailing list
>     >     >>         > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >     >>         <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>>
>     >     >>       
>      >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >     >>         >
>     >     >>
>     >     
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >     >>         > http://www.freeswitch.org
>     >     >>
>     >     >>
>     >
>     _________________________________________________________________________
>     >     >>         Professional FreeSWITCH Consulting Services:
>     >     >> consulting at freeswitch.org
>     <mailto:consulting at freeswitch.org>
>     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>
>     >     <mailto:consulting at freeswitch.org
>     <mailto:consulting at freeswitch.org>
>     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>>
>     >     >>http://www.freeswitchsolutions.com
>     >     >>
>     >     >>         Official FreeSWITCH Sites
>     >     >> http://www.freeswitch.org
>     >     >> http://confluence.freeswitch.org
>     >     >> http://www.cluecon.com
>     >     >>
>     >     >>         FreeSWITCH-users mailing list
>     >     >> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >     >>         <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>>
>     >     >>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >     >>
>     >     
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >     >> http://www.freeswitch.org
>     >     >>
>     >     >>
>     >     >>
>     >     >>
>     >     >>     --
>     >     >>     Sincerely,
>     >     >>
>     >     >>     Giovanni Maruzzelli
>     >     >>     Cell : +39-347-2665618 <tel:%2B39-347-2665618>
>     <tel:%2B39-347-2665618>
>     >     <tel:%2B39-347-2665618>
>     >     >>
>     _________________________________________________________________________
>     >     >>     Professional FreeSWITCH Consulting Services:
>     >     >> consulting at freeswitch.org
>     <mailto:consulting at freeswitch.org>
>     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>
>     >     <mailto:consulting%40freeswitch.org
>     <mailto:consulting%2540freeswitch.org>
>     >     <mailto:consulting%2540freeswitch.org
>     <mailto:consulting%252540freeswitch.org>>>
>     >     >>http://www.freeswitchsolutions.com
>     >     >>
>     >     >>     Official FreeSWITCH Sites
>     >     >> http://www.freeswitch.org
>     >     >> http://confluence.freeswitch.org
>     >     >> http://www.cluecon.com
>     >     >>
>     >     >>     FreeSWITCH-users mailing list
>     >     >> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >     >>     <mailto:FreeSWITCH-users%40lists.freeswitch.org
>     <mailto:FreeSWITCH-users%2540lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users%2540lists.freeswitch.org
>     <mailto:FreeSWITCH-users%252540lists.freeswitch.org>>>
>     >     >>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >     >>
>     >     
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >     >> http://www.freeswitch.org
>     >     >
>     >     >
>     >
>     _________________________________________________________________________
>     >     >     Professional FreeSWITCH Consulting Services:
>     >     > consulting at freeswitch.org
>     <mailto:consulting at freeswitch.org>
>     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>
>     >     <mailto:consulting at freeswitch.org
>     <mailto:consulting at freeswitch.org>
>     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>>
>     >     >http://www.freeswitchsolutions.com
>     >     >
>     >     >     Official FreeSWITCH Sites
>     >     > http://www.freeswitch.org
>     >     > http://confluence.freeswitch.org
>     >     > http://www.cluecon.com
>     >     >
>     >     >     FreeSWITCH-users mailing list
>     >     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>>
>     >     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >     >
>     >     
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >     > http://www.freeswitch.org
>     >     >
>     >     >
>     >     >
>     >     >
>     >
>      _________________________________________________________________________
>     >     > Professional FreeSWITCH Consulting Services:
>     >     > consulting at freeswitch.org
>     <mailto:consulting at freeswitch.org>
>     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>
>     >     > http://www.freeswitchsolutions.com
>     >     >
>     >     > Official FreeSWITCH Sites
>     >     > http://www.freeswitch.org
>     >     > http://confluence.freeswitch.org
>     >     > http://www.cluecon.com
>     >     >
>     >     > FreeSWITCH-users mailing list
>     >     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >     >
>     >   
>      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >     > http://www.freeswitch.org
>     >
>     >
>     >
>     >
>      _________________________________________________________________________
>     >     Professional FreeSWITCH Consulting Services:
>     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>
>     > http://www.freeswitchsolutions.com
>     >
>     >     Official FreeSWITCH Sites
>     > http://www.freeswitch.org
>     > http://confluence.freeswitch.org
>     > http://www.cluecon.com
>     >
>     >     FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >   
>      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>     >
>     >
>     >
>     >
>     > --
>     > Sincerely,
>     >
>     > Giovanni Maruzzelli
>     > Cell : +39-347-2665618 <tel:%2B39-347-2665618>
>     >
>     >
>     >
>     _________________________________________________________________________
>     > Professional FreeSWITCH Consulting Services:
>     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     > http://www.freeswitchsolutions.com
>     >
>     > Official FreeSWITCH Sites
>     > http://www.freeswitch.org
>     > http://confluence.freeswitch.org
>     > http://www.cluecon.com
>     >
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From gmaruzz at gmail.com  Sun Jun  5 18:40:36 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sun, 5 Jun 2016 16:40:36 +0200
Subject: [Freeswitch-users] mod_gsmopen: are USB dongles the only option?
In-Reply-To: <CALXCt0rqW3or4=k5GSjqU4mbt2=D4SRT9Uh4m-80UWV2TOLcaA@mail.gmail.com>
References: <CACOYK=g=DrJipsvG6jg8yqUjC+bd5rhsUfgRH7eOf+tsF14=fQ@mail.gmail.com>
	<CALXCt0qpUzz115iNjBLcTUpWCXZkpooVhL574eTC7zHM+VrDWA@mail.gmail.com>
	<CALXCt0rqW3or4=k5GSjqU4mbt2=D4SRT9Uh4m-80UWV2TOLcaA@mail.gmail.com>
Message-ID: <CALXCt0qVKarWoyS5XGQf5MbRDcN-9Lq0KunJGuk2_B8gcLYaKQ@mail.gmail.com>

Ciao Stanislav,

the whole idea of gsmopen is to use cheap and ready available radio
interfaces, that maybe you can get off ebay or surplus.

Also, they are undistinguishable from "normally used" dongles, and that is
a plus in a lot of countries.

I don't know about pci modems that exposes an audio interface to the
operating system.

Some soldering and demuxing will be probably needed.

But at that point, I would go with Samgoma or Khomp cards...

Also because for doing it well, a lot of engineering details are involved,
from shielding to low pass filtering, to magnetic interferences, etc

Anyway, I like those things too! :)

-giovanni

sent from mobile
cell: +39 347 266 56 18
Giovanni Maruzzelli
OpenTelecom.IT
Il 05/Giu/2016 16:06, "Stanislav Sinyagin" <ssinyagin at gmail.com> ha scritto:

does anyone know a GSM/UMTS/LTE modem that has mini-PCIe interface and
is able to deliver voice signal that can be passed to mod_gsmopen?

The only one that has any kind of voice signal that I found, is Huawei
MU609 and the like. But they deliver PCM voice as a raw signal on
mini-PCIe pins which are marked as "reserved" in the interface
specifications, so normally the boards don't have any wiring for those
pins.

I just don't like the idea of an USB dongle, and there are many
hardware platforms with mini-PCIe interfaces. They work perfectly for
mobile data, but I couldn't find any option for voice.

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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From anthony.minessale at gmail.com  Sun Jun  5 19:16:56 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 5 Jun 2016 10:16:56 -0500
Subject: [Freeswitch-users] ICE issue
In-Reply-To: <CAL+i1s-4wWWnsZ+g+ZnrLdoe=KPAcD7j4Yq6bpNa28c=Ut1uig@mail.gmail.com>
References: <CAL+i1s9-BPaWvmXnsEAmVV24kd_tB-Mz8a8NW7n=mQXvR0gZMQ@mail.gmail.com>
	<CAO_GGAp3pJmk=ixtgyDgg0C0eDVB2OrVd2jcztZ=oKiM1mTGGA@mail.gmail.com>
	<CAL+i1s-4wWWnsZ+g+ZnrLdoe=KPAcD7j4Yq6bpNa28c=Ut1uig@mail.gmail.com>
Message-ID: <CAKbxfG_JfU69X6L9yGU3sjUmhm6j7eOMMSt1FNOwaGm5gk+rdA@mail.gmail.com>

Try latest master as we had a few patches for using ICE with non-webrtc
endpoints.
Also make sure you have apply-candidate-acl params in your sofia profile
defined in the order of preference for IP you want to prefer.


On Sun, Jun 5, 2016 at 6:10 AM, Bastien LE REST <ba.lerest at gmail.com> wrote:

> Hi,
>
> Thank you for your answer. You are true, when I enable a STUN server in my
> PJSIP the calls work fine. I don't understand how the calls can work with
> this configuration through Asterisk and not through FreeSwitch.
>
> Kind regards,
> Bastien.
>
> On 5 June 2016 at 09:16, Volodymyr Fedorov <lexxua at gmail.com> wrote:
>
>> Hi, seems you did not enable STUN server in your PJSIP-client config and
>> you sending only lan ip addresses as ICE-candidates.
>>
>> Br,
>> Volodymyr
>>
>> On Sat, Jun 4, 2016 at 2:48 PM, Bastien LE REST <ba.lerest at gmail.com>
>> wrote:
>>
>>> Hi,
>>>
>>> When I enable ICE using PJSIP, FreeSwitch answers "488 Not Acceptable
>>> Here". When I disable ICE, I don't have any problems, the calls work just
>>> fine. I tried to use another SIP client with ICE and the call works fine.
>>> So I think the problem has to do with PJSIP.
>>>
>>> I'm working on the migration from an Asterisk architecture to a
>>> FreeSwitch architecture. I have to deal with PJSIP and ICE support because
>>> my client is a mobile application who is already published on the stores.
>>>
>>> PJSIP trace: https://pastebin.freeswitch.org/view/ddeaae4b
>>> FreeSwitch trace: https://pastebin.freeswitch.org/view/d380e01d
>>>
>>> Could you please have a look at the trace and tell me what am I doing
>>> wrong? Thank you in advance.
>>>
>>> Kind regards,
>>> Bastien.
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Best regards,
>> Volodymyr
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From 35633 at heb.be  Sun Jun  5 21:58:01 2016
From: 35633 at heb.be (Nduwayezu, Joselyne)
Date: Sun, 5 Jun 2016 19:58:01 +0200
Subject: [Freeswitch-users] Git error
In-Reply-To: <07E9BF9B-A19A-4B0F-8DE3-80C251F5413B@freeswitch.org>
References: <CAFSzUtaM6cR+R6KnOA8NuU63rtPKqytAgnQ7MyFf1i4346rL+A@mail.gmail.com>
	<07E9BF9B-A19A-4B0F-8DE3-80C251F5413B@freeswitch.org>
Message-ID: <CAFSzUtbOkywukQUbR+2wE_Kq-tVj-3bajUTtNfezK5CSoS=67g@mail.gmail.com>

Hello,

The ffirst problem is fixed. I still looking for the second. In fact, i
would like to download music (MOH) i can play when a call hit the dialplan.
Could you suggest me something?

Thank you.

NDUWAYEZU Joselyne

2016-06-04 15:30 GMT+02:00 Ken Rice <krice at freeswitch.org>:

> Find some updated documentation on confluence at freeswitch.org
>
> We dropped support for the git protocol ages ago. The git repo is
> accessible over https or ssh
>
> Sent from my iPhone
>
> On Jun 4, 2016, at 2:43 AM, Nduwayezu, Joselyne <35633 at heb.be> wrote:
>
>
> Hello,
> I'm trying to have clone this one:
>
> *1) git clone git://git.freeswitch.org/freeswitch-contrib.git
> <http://git.freeswitch.org/freeswitch-contrib.git>*
>
> but i have the following error
>
> "fatal: unable to connecte to git.freeswitch.org
>
> git.freeswitch.org[0: 209.105.235.6]: erno=Connection refused
>
> git.freeswitch.og[1: 2607:f348:1021::6]: erno=Network is unreachable
>
> But when i ping git.freeswitch.org or ping 209.105.235.6, the ping success
>
> 2)
>
> *wget http://music.incompetech.com/royalty-free/Parisian.mp3
> <http://music.incompetech.com/royalty-free/Parisian.mp3> *
>
> the error is:
>
> HTTP request sent, awaiting response ....  404 Not Found
>
> Any ideas? Thanks
>
> NDUWAYEZU Joselyne
>
> Haute ?cole de Bruxelles
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Haute ?cole de Bruxelles
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From bipin at xbipin.com  Mon Jun  6 00:07:22 2016
From: bipin at xbipin.com (Bipin Patel)
Date: Mon, 06 Jun 2016 00:07:22 +0400
Subject: [Freeswitch-users] Inbound calls from GSM to extension that called
	person on gsm
Message-ID: <155222d48a8.2787.b07ebdf329620b8089087c7205b03f01@xbipin.com>

I'm having one client who has a special requirement, i have setup FS for 
them using static XML files and a GSM dongle, they are able to call other 
extensions and gateways as well as dial out of the GSM line but they wanted 
incoming calls from GSM to ring the extension that called the person using 
the GSM line. Meaning extension 1008 calls a number which FS routes out of 
the GSM line so the called party sees the GSM number always and suppose if 
the called party is busy and doesn't answer or rejects call then when later 
he dials back in I want FS to know which extension had called that number 
so incoming calls from the GSM would ring that extension only.

I know I can do this dynamically using XML curl and CDR etc but is there a 
simple way to achieve this using static XML files only as this FS setup is 
very small and very basic





From mike at jerris.com  Mon Jun  6 01:00:57 2016
From: mike at jerris.com (Michael Jerris)
Date: Sun, 5 Jun 2016 17:00:57 -0400
Subject: [Freeswitch-users] Git error
In-Reply-To: <CAFSzUtbOkywukQUbR+2wE_Kq-tVj-3bajUTtNfezK5CSoS=67g@mail.gmail.com>
References: <CAFSzUtaM6cR+R6KnOA8NuU63rtPKqytAgnQ7MyFf1i4346rL+A@mail.gmail.com>
	<07E9BF9B-A19A-4B0F-8DE3-80C251F5413B@freeswitch.org>
	<CAFSzUtbOkywukQUbR+2wE_Kq-tVj-3bajUTtNfezK5CSoS=67g@mail.gmail.com>
Message-ID: <CAEwTsAhz-1kDa4gH35VYbAmqt73HOzwCcsUbG+NDMHbTs5o6MQ@mail.gmail.com>

The other issue seems to be a link to someone else's music, we would have
no idea what is wrong with that.  You are more than welcome to get the
music we package or any other music you like.

On Sunday, June 5, 2016, Nduwayezu, Joselyne <35633 at heb.be> wrote:

> Hello,
>
> The ffirst problem is fixed. I still looking for the second. In fact, i
> would like to download music (MOH) i can play when a call hit the dialplan.
> Could you suggest me something?
>
> Thank you.
>
> NDUWAYEZU Joselyne
>
> 2016-06-04 15:30 GMT+02:00 Ken Rice <krice at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','krice at freeswitch.org');>>:
>
>> Find some updated documentation on confluence at freeswitch.org
>>
>> We dropped support for the git protocol ages ago. The git repo is
>> accessible over https or ssh
>>
>> Sent from my iPhone
>>
>> On Jun 4, 2016, at 2:43 AM, Nduwayezu, Joselyne <35633 at heb.be
>> <javascript:_e(%7B%7D,'cvml','35633 at heb.be');>> wrote:
>>
>>
>> Hello,
>> I'm trying to have clone this one:
>>
>> *1) git clone git://git.freeswitch.org/freeswitch-contrib.git
>> <http://git.freeswitch.org/freeswitch-contrib.git>*
>>
>> but i have the following error
>>
>> "fatal: unable to connecte to git.freeswitch.org
>>
>> git.freeswitch.org[0: 209.105.235.6]: erno=Connection refused
>>
>> git.freeswitch.og[1: 2607:f348:1021::6]: erno=Network is unreachable
>>
>> But when i ping git.freeswitch.org or ping 209.105.235.6, the ping
>> success
>>
>> 2)
>>
>> *wget http://music.incompetech.com/royalty-free/Parisian.mp3
>> <http://music.incompetech.com/royalty-free/Parisian.mp3> *
>>
>> the error is:
>>
>> HTTP request sent, awaiting response ....  404 Not Found
>>
>> Any ideas? Thanks
>>
>> NDUWAYEZU Joselyne
>>
>> Haute ?cole de Bruxelles
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> Haute ?cole de Bruxelles
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From ssinyagin at gmail.com  Mon Jun  6 03:30:25 2016
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Mon, 6 Jun 2016 01:30:25 +0200
Subject: [Freeswitch-users] mod_gsmopen: are USB dongles the only option?
In-Reply-To: <CALXCt0qVKarWoyS5XGQf5MbRDcN-9Lq0KunJGuk2_B8gcLYaKQ@mail.gmail.com>
References: <CACOYK=g=DrJipsvG6jg8yqUjC+bd5rhsUfgRH7eOf+tsF14=fQ@mail.gmail.com>
	<CALXCt0qpUzz115iNjBLcTUpWCXZkpooVhL574eTC7zHM+VrDWA@mail.gmail.com>
	<CALXCt0rqW3or4=k5GSjqU4mbt2=D4SRT9Uh4m-80UWV2TOLcaA@mail.gmail.com>
	<CALXCt0qVKarWoyS5XGQf5MbRDcN-9Lq0KunJGuk2_B8gcLYaKQ@mail.gmail.com>
Message-ID: <CACOYK=i03TXCcrLyLDtoy7g7gjtjSzWCCrX42GCbTTPaJsD8Bg@mail.gmail.com>

those mPCIe 3G modems that I tried, are all visible as USB devices,
with serial (ttyUSB) and (when supported) Ethernet devices associated
with them.

So, in theory, the audio signal would also be available in the same
way as with the USB dongle... but so far, I haven't seen such a device
yet.



On Sun, Jun 5, 2016 at 4:40 PM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> Ciao Stanislav,
>
> the whole idea of gsmopen is to use cheap and ready available radio
> interfaces, that maybe you can get off ebay or surplus.
>
> Also, they are undistinguishable from "normally used" dongles, and that is a
> plus in a lot of countries.
>
> I don't know about pci modems that exposes an audio interface to the
> operating system.
>
> Some soldering and demuxing will be probably needed.
>
> But at that point, I would go with Samgoma or Khomp cards...
>
> Also because for doing it well, a lot of engineering details are involved,
> from shielding to low pass filtering, to magnetic interferences, etc
>
> Anyway, I like those things too! :)
>
> -giovanni
>
> sent from mobile
> cell: +39 347 266 56 18
> Giovanni Maruzzelli
> OpenTelecom.IT
>
> Il 05/Giu/2016 16:06, "Stanislav Sinyagin" <ssinyagin at gmail.com> ha scritto:
>
> does anyone know a GSM/UMTS/LTE modem that has mini-PCIe interface and
> is able to deliver voice signal that can be passed to mod_gsmopen?
>
> The only one that has any kind of voice signal that I found, is Huawei
> MU609 and the like. But they deliver PCM voice as a raw signal on
> mini-PCIe pins which are marked as "reserved" in the interface
> specifications, so normally the boards don't have any wiring for those
> pins.
>
> I just don't like the idea of an USB dongle, and there are many
> hardware platforms with mini-PCIe interfaces. They work perfectly for
> mobile data, but I couldn't find any option for voice.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From ba.lerest at gmail.com  Mon Jun  6 12:58:52 2016
From: ba.lerest at gmail.com (Bastien LE REST)
Date: Mon, 6 Jun 2016 09:58:52 +0100
Subject: [Freeswitch-users] ICE issue
In-Reply-To: <CAKbxfG_JfU69X6L9yGU3sjUmhm6j7eOMMSt1FNOwaGm5gk+rdA@mail.gmail.com>
References: <CAL+i1s9-BPaWvmXnsEAmVV24kd_tB-Mz8a8NW7n=mQXvR0gZMQ@mail.gmail.com>
	<CAO_GGAp3pJmk=ixtgyDgg0C0eDVB2OrVd2jcztZ=oKiM1mTGGA@mail.gmail.com>
	<CAL+i1s-4wWWnsZ+g+ZnrLdoe=KPAcD7j4Yq6bpNa28c=Ut1uig@mail.gmail.com>
	<CAKbxfG_JfU69X6L9yGU3sjUmhm6j7eOMMSt1FNOwaGm5gk+rdA@mail.gmail.com>
Message-ID: <CAL+i1s-pWA+wOBxFyHLG0+6+MJKhzm9zn63S6GN0gJw1BV3d3A@mail.gmail.com>

Hi,

Merci Anthony! It's working now. It was an issue with the ACL.

Kind regards,
Bastien.



On 5 June 2016 at 16:16, Anthony Minessale <anthony.minessale at gmail.com>
wrote:

> Try latest master as we had a few patches for using ICE with non-webrtc
> endpoints.
> Also make sure you have apply-candidate-acl params in your sofia profile
> defined in the order of preference for IP you want to prefer.
>
>
> On Sun, Jun 5, 2016 at 6:10 AM, Bastien LE REST <ba.lerest at gmail.com>
> wrote:
>
>> Hi,
>>
>> Thank you for your answer. You are true, when I enable a STUN server in
>> my PJSIP the calls work fine. I don't understand how the calls can work
>> with this configuration through Asterisk and not through FreeSwitch.
>>
>> Kind regards,
>> Bastien.
>>
>> On 5 June 2016 at 09:16, Volodymyr Fedorov <lexxua at gmail.com> wrote:
>>
>>> Hi, seems you did not enable STUN server in your PJSIP-client config and
>>> you sending only lan ip addresses as ICE-candidates.
>>>
>>> Br,
>>> Volodymyr
>>>
>>> On Sat, Jun 4, 2016 at 2:48 PM, Bastien LE REST <ba.lerest at gmail.com>
>>> wrote:
>>>
>>>> Hi,
>>>>
>>>> When I enable ICE using PJSIP, FreeSwitch answers "488 Not Acceptable
>>>> Here". When I disable ICE, I don't have any problems, the calls work just
>>>> fine. I tried to use another SIP client with ICE and the call works fine.
>>>> So I think the problem has to do with PJSIP.
>>>>
>>>> I'm working on the migration from an Asterisk architecture to a
>>>> FreeSwitch architecture. I have to deal with PJSIP and ICE support because
>>>> my client is a mobile application who is already published on the stores.
>>>>
>>>> PJSIP trace: https://pastebin.freeswitch.org/view/ddeaae4b
>>>> FreeSwitch trace: https://pastebin.freeswitch.org/view/d380e01d
>>>>
>>>> Could you please have a look at the trace and tell me what am I doing
>>>> wrong? Thank you in advance.
>>>>
>>>> Kind regards,
>>>> Bastien.
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Best regards,
>>> Volodymyr
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Mon Jun  6 16:56:41 2016
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 6 Jun 2016 13:56:41 +0100
Subject: [Freeswitch-users] mod_gsmopen: are USB dongles the only option?
In-Reply-To: <CACOYK=i03TXCcrLyLDtoy7g7gjtjSzWCCrX42GCbTTPaJsD8Bg@mail.gmail.com>
References: <CACOYK=g=DrJipsvG6jg8yqUjC+bd5rhsUfgRH7eOf+tsF14=fQ@mail.gmail.com>
	<CALXCt0qpUzz115iNjBLcTUpWCXZkpooVhL574eTC7zHM+VrDWA@mail.gmail.com>
	<CALXCt0rqW3or4=k5GSjqU4mbt2=D4SRT9Uh4m-80UWV2TOLcaA@mail.gmail.com>
	<CALXCt0qVKarWoyS5XGQf5MbRDcN-9Lq0KunJGuk2_B8gcLYaKQ@mail.gmail.com>
	<CACOYK=i03TXCcrLyLDtoy7g7gjtjSzWCCrX42GCbTTPaJsD8Bg@mail.gmail.com>
Message-ID: <CAFiqYuk5zTtn7=EANngMiBzxT=eXrmwwPn=d_PRn7x2MKmTUYw@mail.gmail.com>

Might be you need to enable phone calls via an AT command.

On 6 June 2016 at 00:30, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

> those mPCIe 3G modems that I tried, are all visible as USB devices,
> with serial (ttyUSB) and (when supported) Ethernet devices associated
> with them.
>
> So, in theory, the audio signal would also be available in the same
> way as with the USB dongle... but so far, I haven't seen such a device
> yet.
>
>
>
> On Sun, Jun 5, 2016 at 4:40 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
> > Ciao Stanislav,
> >
> > the whole idea of gsmopen is to use cheap and ready available radio
> > interfaces, that maybe you can get off ebay or surplus.
> >
> > Also, they are undistinguishable from "normally used" dongles, and that
> is a
> > plus in a lot of countries.
> >
> > I don't know about pci modems that exposes an audio interface to the
> > operating system.
> >
> > Some soldering and demuxing will be probably needed.
> >
> > But at that point, I would go with Samgoma or Khomp cards...
> >
> > Also because for doing it well, a lot of engineering details are
> involved,
> > from shielding to low pass filtering, to magnetic interferences, etc
> >
> > Anyway, I like those things too! :)
> >
> > -giovanni
> >
> > sent from mobile
> > cell: +39 347 266 56 18
> > Giovanni Maruzzelli
> > OpenTelecom.IT
> >
> > Il 05/Giu/2016 16:06, "Stanislav Sinyagin" <ssinyagin at gmail.com> ha
> scritto:
> >
> > does anyone know a GSM/UMTS/LTE modem that has mini-PCIe interface and
> > is able to deliver voice signal that can be passed to mod_gsmopen?
> >
> > The only one that has any kind of voice signal that I found, is Huawei
> > MU609 and the like. But they deliver PCM voice as a raw signal on
> > mini-PCIe pins which are marked as "reserved" in the interface
> > specifications, so normally the boards don't have any wiring for those
> > pins.
> >
> > I just don't like the idea of an USB dongle, and there are many
> > hardware platforms with mini-PCIe interfaces. They work perfectly for
> > mobile data, but I couldn't find any option for voice.
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ssinyagin at gmail.com  Mon Jun  6 18:49:34 2016
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Mon, 6 Jun 2016 16:49:34 +0200
Subject: [Freeswitch-users] mod_gsmopen: are USB dongles the only option?
In-Reply-To: <CAFiqYuk5zTtn7=EANngMiBzxT=eXrmwwPn=d_PRn7x2MKmTUYw@mail.gmail.com>
References: <CACOYK=g=DrJipsvG6jg8yqUjC+bd5rhsUfgRH7eOf+tsF14=fQ@mail.gmail.com>
	<CALXCt0qpUzz115iNjBLcTUpWCXZkpooVhL574eTC7zHM+VrDWA@mail.gmail.com>
	<CALXCt0rqW3or4=k5GSjqU4mbt2=D4SRT9Uh4m-80UWV2TOLcaA@mail.gmail.com>
	<CALXCt0qVKarWoyS5XGQf5MbRDcN-9Lq0KunJGuk2_B8gcLYaKQ@mail.gmail.com>
	<CACOYK=i03TXCcrLyLDtoy7g7gjtjSzWCCrX42GCbTTPaJsD8Bg@mail.gmail.com>
	<CAFiqYuk5zTtn7=EANngMiBzxT=eXrmwwPn=d_PRn7x2MKmTUYw@mail.gmail.com>
Message-ID: <CACOYK=jfDnc9S63PMEje_CqZBYLJLsOi+szCuuYwANcXhLv2Mw@mail.gmail.com>

no, it's the physical capability of particular modem. Some of them
don;t have voice call capability at all. Some of them do have, and the
call is sent to the pins which are not wired on your PCB :)




On Mon, Jun 6, 2016 at 2:56 PM, Steven Ayre <steveayre at gmail.com> wrote:
> Might be you need to enable phone calls via an AT command.
>
> On 6 June 2016 at 00:30, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
>>
>> those mPCIe 3G modems that I tried, are all visible as USB devices,
>> with serial (ttyUSB) and (when supported) Ethernet devices associated
>> with them.
>>
>> So, in theory, the audio signal would also be available in the same
>> way as with the USB dongle... but so far, I haven't seen such a device
>> yet.
>>
>>
>>
>> On Sun, Jun 5, 2016 at 4:40 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>> wrote:
>> > Ciao Stanislav,
>> >
>> > the whole idea of gsmopen is to use cheap and ready available radio
>> > interfaces, that maybe you can get off ebay or surplus.
>> >
>> > Also, they are undistinguishable from "normally used" dongles, and that
>> > is a
>> > plus in a lot of countries.
>> >
>> > I don't know about pci modems that exposes an audio interface to the
>> > operating system.
>> >
>> > Some soldering and demuxing will be probably needed.
>> >
>> > But at that point, I would go with Samgoma or Khomp cards...
>> >
>> > Also because for doing it well, a lot of engineering details are
>> > involved,
>> > from shielding to low pass filtering, to magnetic interferences, etc
>> >
>> > Anyway, I like those things too! :)
>> >
>> > -giovanni
>> >
>> > sent from mobile
>> > cell: +39 347 266 56 18
>> > Giovanni Maruzzelli
>> > OpenTelecom.IT
>> >
>> > Il 05/Giu/2016 16:06, "Stanislav Sinyagin" <ssinyagin at gmail.com> ha
>> > scritto:
>> >
>> > does anyone know a GSM/UMTS/LTE modem that has mini-PCIe interface and
>> > is able to deliver voice signal that can be passed to mod_gsmopen?
>> >
>> > The only one that has any kind of voice signal that I found, is Huawei
>> > MU609 and the like. But they deliver PCM voice as a raw signal on
>> > mini-PCIe pins which are marked as "reserved" in the interface
>> > specifications, so normally the boards don't have any wiring for those
>> > pins.
>> >
>> > I just don't like the idea of an USB dongle, and there are many
>> > hardware platforms with mini-PCIe interfaces. They work perfectly for
>> > mobile data, but I couldn't find any option for voice.
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From trever at middleearth.sapphiresunday.org  Mon Jun  6 22:54:37 2016
From: trever at middleearth.sapphiresunday.org (Trever L. Adams)
Date: Mon, 6 Jun 2016 12:54:37 -0600
Subject: [Freeswitch-users] Silence Detection in Recording
In-Reply-To: <CADkd9g0jhPOrucerx9jpGnme0BCNTdZWk4FCVTrUM2gDoX_nQA@mail.gmail.com>
References: <CADkd9g0jhPOrucerx9jpGnme0BCNTdZWk4FCVTrUM2gDoX_nQA@mail.gmail.com>
Message-ID: <35a8cce1-b6ba-ca16-afe7-427f8df73b48@middleearth.sapphiresunday.org>

On 05/13/2016 01:39 AM, Jeevan Singh Dhami wrote:
> Hi,
> I have  recorded the  voice using freeswitch and  their found silence
> in recording if recording goes below  /given silence threshold. after
> that freeswitch stop the recording  .but there was no indication given
> by freeswitch for silence detection .Could  you please tell me there
> is any way to find the indication parameters in freeswitch. Either
> any solution for knowing silence found in recording in freeswitch./
> /please share the solution in same mail .i will appreciate your
> feedback. /
It sounds to me like you are using record app in FreeSWITCH, it stops
recording due to timeout or silence detection. And you are wanting to
know if the entire file was silence?

If I have understood you correctly, I have not been able to find a way
to do this directly in FreeSWITCH. I use a bash wrapper around sox to
extract the RMS delta level. If it exceeds a certain level (I have found
what largely works for my phone system, but it will be different for
yours I am sure), then there is actual usable audio. FreeSWITCH can call
external programs like that. I needed my shell script for a lua based
call screener. (I didn't want background noise and pops and clicks to be
a reason to ring the internal phones.)

I hope this helps,
Trever
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From trever at middleearth.sapphiresunday.org  Tue Jun  7 01:50:19 2016
From: trever at middleearth.sapphiresunday.org (Trever L. Adams)
Date: Mon, 6 Jun 2016 15:50:19 -0600
Subject: [Freeswitch-users] calling all native speakers of languages
 other than english. We need your help, and this will be easy.
In-Reply-To: <391E7162-7420-4CF9-A229-936DC3A5636D@jerris.com>
References: <391E7162-7420-4CF9-A229-936DC3A5636D@jerris.com>
Message-ID: <f11ee79d-327f-c6d1-d8a8-dfede83d9a8f@middleearth.sapphiresunday.org>

On 03/29/2016 02:15 PM, Michael Jerris wrote:
> Just committed into tree is internationalization of Verto
> Communicator, so far, just English and Italian.  If you speak another
> language, we need your help to translate these into other languages.
>  The file that needs to be translated is:
>
> https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/html5/verto/verto_communicator/src/locales/locale-en.json?at=6c197ae2f0cb647ab20bc5143870a80c3dd8d602&raw
>
> If you can assist with this, Please make a new file named
> locale-language.json (where "language" is the 2 letter code for your
> language) and translate all the text in quotes to the right thing for
> your language and send them back our way.  Lets see how quickly we can
> crowd source this
>
> Thanks!
I am not a native speaker of Hebrew, but I have been learning for
several years and can read, write and speak like a child (and more
complex documents such as the Tanakh). I cannot say this is error free,
but it is MUCH better than what a machine translation would likely be. I
am not seeing these files in my local repository for some reason.

If you are interested, here it is.

Thank you,
Trever
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From marcel.haldemann at convercom.ch  Tue Jun  7 12:44:50 2016
From: marcel.haldemann at convercom.ch (Marcel Haldemann)
Date: Tue, 7 Jun 2016 08:44:50 +0000
Subject: [Freeswitch-users] mod_H.323, working stable ?
Message-ID: <VI1PR06MB131106332C9A7751D6917B16EF5D0@VI1PR06MB1311.eurprd06.prod.outlook.com>

Howdy fellow Freeswich-Users,

We are using FreeSWITCH since while now with SIP. However, recently I got ask what I would take to support H.323.
I then checked the Docs and the commits in the git repos for mod_h323 and mod_opal.
The changes in the repos seems to be similar (not mutch (only some VS2015 updates) going on in recent years). Regarding the Docs it seems better to use mod_h323 rather than  mod_opal, as it is in a "beta" stage (info from 2008, maybe outdated ?).
The underlying project H323plus seems to have stopped development in 2014.

Is there someone, currently using H.323 with FreeSwitch and made it working stable (preferably using FS 1.6.x) ?
If so, any hints on what mod to use (mod_h323 or mod_opal) ? any traps ? regarding docs in the H323 mod the trunk version of PTLIB does not work. What Version did u use ?

I think it would be better to use mod_h323 or is it suggested to use mod_opal ?

What are your experiences in H.323 with FreeSwitch, does it work stable ?

Trying to get an impression whether it's a good idea to use the H.323 mod (or opal), I'm happy about any feedback. Also if someone countered unresolvable problems with the H.323 mod.

Kind regards


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From matt at supportedbusiness.com  Tue Jun  7 14:20:31 2016
From: matt at supportedbusiness.com (Matt Broad)
Date: Tue, 7 Jun 2016 11:20:31 +0100
Subject: [Freeswitch-users] add modules when installing from the latest
	branch
Message-ID: <CAEeV61jv_rX-ebF6O4gM3=DG3jEDdMaBBovYtj_2QAwzozPJrg@mail.gmail.com>

Hi,

this may be a really stupid question but here goes:

I have upgraded a server to debian 8 and have installed freeswitch using
the latest branch
https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie#Debian8Jessie-latestreleasebranch
:

My question is how do I activate other modules?  In previous versions I
would have amended the moudules conf file and re-compile.  I can't find the
modules file in this install


thanks
Matt
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From benjamin.cropley at gmail.com  Tue Jun  7 14:35:49 2016
From: benjamin.cropley at gmail.com (Benjamin Cropley)
Date: Tue, 7 Jun 2016 11:35:49 +0100
Subject: [Freeswitch-users] add modules when installing from the latest
	branch
In-Reply-To: <CAEeV61jv_rX-ebF6O4gM3=DG3jEDdMaBBovYtj_2QAwzozPJrg@mail.gmail.com>
References: <CAEeV61jv_rX-ebF6O4gM3=DG3jEDdMaBBovYtj_2QAwzozPJrg@mail.gmail.com>
Message-ID: <CAF7jizJ7yXS5a0QhFm5uY1vZttcmO5HoM3sssCAqLY1NJyO37g@mail.gmail.com>

Say you wanted to install mod_mongo..

you can find the packages here
http://files.freeswitch.org/repo/deb/freeswitch-1.6/dists/jessie/main/binary-amd64/Packages

apt-get install -y freeswitch-mod-mongo

then edit the modules conf as you would have done previously

On Tue, Jun 7, 2016 at 11:20 AM, Matt Broad <matt at supportedbusiness.com>
wrote:

> Hi,
>
> this may be a really stupid question but here goes:
>
> I have upgraded a server to debian 8 and have installed freeswitch using
> the latest branch
>
> https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie#Debian8Jessie-latestreleasebranch
> :
>
> My question is how do I activate other modules?  In previous versions I
> would have amended the moudules conf file and re-compile.  I can't find the
> modules file in this install
>
>
> thanks
> Matt
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From matt at supportedbusiness.com  Tue Jun  7 14:49:48 2016
From: matt at supportedbusiness.com (Matt Broad)
Date: Tue, 7 Jun 2016 11:49:48 +0100
Subject: [Freeswitch-users] add modules when installing from the latest
	branch
In-Reply-To: <CAF7jizJ7yXS5a0QhFm5uY1vZttcmO5HoM3sssCAqLY1NJyO37g@mail.gmail.com>
References: <CAEeV61jv_rX-ebF6O4gM3=DG3jEDdMaBBovYtj_2QAwzozPJrg@mail.gmail.com>
	<CAF7jizJ7yXS5a0QhFm5uY1vZttcmO5HoM3sssCAqLY1NJyO37g@mail.gmail.com>
Message-ID: <CAEeV61g12Ar2euivZQb+3NkTB7mRO7keRe12AN7kp9J8Do3xZw@mail.gmail.com>

perfect!

thanks for the quick response.  Is this listed on the confluence site
anywhere?  I am happy to add this if given access

thanks
Matt

On 7 June 2016 at 11:35, Benjamin Cropley <benjamin.cropley at gmail.com>
wrote:

> Say you wanted to install mod_mongo..
>
> you can find the packages here
> http://files.freeswitch.org/repo/deb/freeswitch-1.6/dists/jessie/main/binary-amd64/Packages
>
> apt-get install -y freeswitch-mod-mongo
>
> then edit the modules conf as you would have done previously
>
> On Tue, Jun 7, 2016 at 11:20 AM, Matt Broad <matt at supportedbusiness.com>
> wrote:
>
>> Hi,
>>
>> this may be a really stupid question but here goes:
>>
>> I have upgraded a server to debian 8 and have installed freeswitch using
>> the latest branch
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie#Debian8Jessie-latestreleasebranch
>> :
>>
>> My question is how do I activate other modules?  In previous versions I
>> would have amended the moudules conf file and re-compile.  I can't find the
>> modules file in this install
>>
>>
>> thanks
>> Matt
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From benjamin.cropley at gmail.com  Tue Jun  7 15:26:39 2016
From: benjamin.cropley at gmail.com (Benjamin Cropley)
Date: Tue, 7 Jun 2016 12:26:39 +0100
Subject: [Freeswitch-users] add modules when installing from the latest
	branch
In-Reply-To: <CAEeV61g12Ar2euivZQb+3NkTB7mRO7keRe12AN7kp9J8Do3xZw@mail.gmail.com>
References: <CAEeV61jv_rX-ebF6O4gM3=DG3jEDdMaBBovYtj_2QAwzozPJrg@mail.gmail.com>
	<CAF7jizJ7yXS5a0QhFm5uY1vZttcmO5HoM3sssCAqLY1NJyO37g@mail.gmail.com>
	<CAEeV61g12Ar2euivZQb+3NkTB7mRO7keRe12AN7kp9J8Do3xZw@mail.gmail.com>
Message-ID: <CAF7jiz+z9TNW1F7OB+ydXNyMPVP_K7zGYkkudi7v3sP=LKpxBQ@mail.gmail.com>

To be honest I'm not sure it's entirely clear this is how you do it on
confluence.. In that theres no "How do I install x module".

To get access you need to join the freeswitch-docs mailing list and drop an
email there intro?ing yourself, one of the lead editors should get your
access fixed up. :)

Ben



On Tue, Jun 7, 2016 at 11:49 AM, Matt Broad <matt at supportedbusiness.com>
wrote:

> perfect!
>
> thanks for the quick response.  Is this listed on the confluence site
> anywhere?  I am happy to add this if given access
>
> thanks
> Matt
>
> On 7 June 2016 at 11:35, Benjamin Cropley <benjamin.cropley at gmail.com>
> wrote:
>
>> Say you wanted to install mod_mongo..
>>
>> you can find the packages here
>> http://files.freeswitch.org/repo/deb/freeswitch-1.6/dists/jessie/main/binary-amd64/Packages
>>
>> apt-get install -y freeswitch-mod-mongo
>>
>> then edit the modules conf as you would have done previously
>>
>> On Tue, Jun 7, 2016 at 11:20 AM, Matt Broad <matt at supportedbusiness.com>
>> wrote:
>>
>>> Hi,
>>>
>>> this may be a really stupid question but here goes:
>>>
>>> I have upgraded a server to debian 8 and have installed freeswitch using
>>> the latest branch
>>>
>>> https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie#Debian8Jessie-latestreleasebranch
>>> :
>>>
>>> My question is how do I activate other modules?  In previous versions I
>>> would have amended the moudules conf file and re-compile.  I can't find the
>>> modules file in this install
>>>
>>>
>>> thanks
>>> Matt
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From pashdown at xmission.com  Tue Jun  7 18:50:38 2016
From: pashdown at xmission.com (Pete Ashdown)
Date: Tue, 7 Jun 2016 08:50:38 -0600
Subject: [Freeswitch-users] System global mode variable
Message-ID: <5756DF3E.8000004@xmission.com>

What is the simplest way to have a conditional branch based on a global
variable in a dialplan?  Actually using a global variable and xml is
giving me mixed results.  Do I need to use Lua or another scripting
language?

This is what I have, which doesn't give the results desired.   The
"speak" actions are merely debug placeholders.  I know there are better
ways to do conditional voice to text.

      <condition regex="any" field="${cond(${mode} == 'open' ? YES :
NO)}" expression="^YES$">
            <action application="speak" data="open"/>
      </condition>

      <condition regex="any" field="${cond(${mode} == 'tech only' ? YES
: NO)}" expression="^YES$">
            <action application="speak" data="teck only"/>
      </condition>

      <condition regex="any" field="${cond(${mode} == 'event' ? YES :
NO)}" expression="^YES$">
            <action application="speak" data="event"/>
      </condition>



From deepikay at iiitd.ac.in  Tue Jun  7 09:20:52 2016
From: deepikay at iiitd.ac.in (Deepika Yadav)
Date: Tue, 7 Jun 2016 10:50:52 +0530
Subject: [Freeswitch-users] GSM gateway to Freeswitch audio problem
Message-ID: <CAM99_Ym407j+aXehJDgYkeTpgX7UDZHn4sdt_SXEi87bkHzatg@mail.gmail.com>

Hi,

I want to make calls forwarded from GSM gateway to Freeswitch extension

When I dial a 10- digit mobile number of a sim configured in the GSM
gateway in a way that it forwards the incoming call to freeswitch
extension. I do not hear anything except for some background noise

Freeswitch codec configuration :

 <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=GSM,PCMU,PCMA"/>

GSM- Gateway Codec settings :

priority 1 - G.711 u-law
priority 2 - G.711 a-law

Dialplan in public context :

<include>
  <extension name="public_did">
    <condition field="destination_number" expression="^(9811806537)$">
        <action application="transfer" data="5000 XML default"/>
    </condition>
  </extension>
</include>



Freeswitch logs:

2016-06-07 05:10:51.000656 [NOTICE] switch_channel.c:1055 New Channel
sofia/external/00919716517818 at 192.168.20.36:5060
[33e4e958-2c6e-11e6-9af0-278859dd14e6]
2016-06-07 05:10:51.000656 [INFO] mod_dialplan_xml.c:635 Processing
00919716517818 <00919716517818>->9811806537 in context public
2016-06-07 05:10:51.000656 [NOTICE] switch_ivr.c:1861 Transfer
sofia/external/00919716517818 at 192.168.20.36:5060 to XML[5000 at default]
2016-06-07 05:10:51.000656 [INFO] mod_dialplan_xml.c:635 Processing
00919716517818 <00919716517818>->5000 in context default
2016-06-07 05:10:51.000656 [NOTICE] sofia_media.c:92 Pre-Answer
sofia/external/00919716517818 at 192.168.20.36:5060!
2016-06-07 05:10:51.000656 [NOTICE] mod_dptools.c:1268 Channel
[sofia/external/00919716517818 at 192.168.20.36:5060] has been answered
2016-06-07 05:10:51.640675 [INFO] switch_rtp.c:5846 Auto Changing port from
192.168.20.36:31012 to 103.25.231.101:31012
2016-06-07 05:11:06.560680 [NOTICE] sofia.c:952 Hangup sofia/external/
00919716517818 at 192.168.20.36:5060 [CS_EXECUTE] [NORMAL_CLEARING]
2016-06-07 05:11:06.560680 [NOTICE] switch_core_session.c:1641 Session 262
(sofia/external/00919716517818 at 192.168.20.36:5060) Ended
2016-06-07 05:11:06.560680 [NOTICE] switch_core_session.c:1645 Close
Channel sofia/external/00919716517818 at 192.168.20.36:5060 [CS_DESTROY]


Regards,
Deepika
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From krice at freeswitch.org  Tue Jun  7 19:35:13 2016
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 7 Jun 2016 10:35:13 -0500
Subject: [Freeswitch-users] add modules when installing from the
	latest	branch
In-Reply-To: <CAEeV61g12Ar2euivZQb+3NkTB7mRO7keRe12AN7kp9J8Do3xZw@mail.gmail.com>
References: <CAEeV61jv_rX-ebF6O4gM3=DG3jEDdMaBBovYtj_2QAwzozPJrg@mail.gmail.com>	<CAF7jizJ7yXS5a0QhFm5uY1vZttcmO5HoM3sssCAqLY1NJyO37g@mail.gmail.com>
	<CAEeV61g12Ar2euivZQb+3NkTB7mRO7keRe12AN7kp9J8Do3xZw@mail.gmail.com>
Message-ID: <007501d1c0d2$2febc220$8fc34660$@freeswitch.org>

Easier is probably just apt-cache search freeswitch or apt-cache search freeswitch|grep mongo or something like that

 

Once you do that you can apt-get install the module you want and then you just need to enable it in your /etc/freeswitch/autoload_configs/modules.conf.xml

 

The entire package system is designed so that you can use https://freeswitch.org/stash/projects/FS/repos/freeswitch-config-skel/browse to generate your own configuration package with deps on the modules that you want to include and just install that and bam you?re good

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Matt Broad
Sent: Tuesday, June 7, 2016 5:50 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] add modules when installing from the latest branch

 

perfect! 

 

thanks for the quick response.  Is this listed on the confluence site anywhere?  I am happy to add this if given access

 

thanks

Matt

 

On 7 June 2016 at 11:35, Benjamin Cropley <benjamin.cropley at gmail.com <mailto:benjamin.cropley at gmail.com> > wrote:

Say you wanted to install mod_mongo..

 

you can find the packages here http://files.freeswitch.org/repo/deb/freeswitch-1.6/dists/jessie/main/binary-amd64/Packages

 

apt-get install -y freeswitch-mod-mongo

 

then edit the modules conf as you would have done previously

 

On Tue, Jun 7, 2016 at 11:20 AM, Matt Broad <matt at supportedbusiness.com <mailto:matt at supportedbusiness.com> > wrote:

Hi,

 

this may be a really stupid question but here goes:

 

I have upgraded a server to debian 8 and have installed freeswitch using the latest branch

https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie#Debian8Jessie-latestreleasebranch:

 

My question is how do I activate other modules?  In previous versions I would have amended the moudules conf file and re-compile.  I can't find the modules file in this install 

 

 

thanks

Matt

 

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

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

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

 


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

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

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

 

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From gmaruzz at gmail.com  Tue Jun  7 19:43:36 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 7 Jun 2016 17:43:36 +0200
Subject: [Freeswitch-users] GSM gateway to Freeswitch audio problem
In-Reply-To: <CAM99_Ym407j+aXehJDgYkeTpgX7UDZHn4sdt_SXEi87bkHzatg@mail.gmail.com>
References: <CAM99_Ym407j+aXehJDgYkeTpgX7UDZHn4sdt_SXEi87bkHzatg@mail.gmail.com>
Message-ID: <CALXCt0p5+GhCJycbSrx0+AA6-PL4tzjcUjvcbcowimak3ndW1Q@mail.gmail.com>

Delete gsm from.codecs (both inbound and outbound) and try again.

Also, activate debug on console, so you can see more of what happens.

Read the troubleshooting and debugging techniques from
www.freeswitch.org/confluence

sent from mobile
cell: +39 347 266 56 18
Giovanni Maruzzelli
OpenTelecom.IT
Il 07/Giu/2016 17:32, "Deepika Yadav" <deepikay at iiitd.ac.in> ha scritto:

> Hi,
>
> I want to make calls forwarded from GSM gateway to Freeswitch extension
>
> When I dial a 10- digit mobile number of a sim configured in the GSM
> gateway in a way that it forwards the incoming call to freeswitch
> extension. I do not hear anything except for some background noise
>
> Freeswitch codec configuration :
>
>  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA"/>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=GSM,PCMU,PCMA"/>
>
> GSM- Gateway Codec settings :
>
> priority 1 - G.711 u-law
> priority 2 - G.711 a-law
>
> Dialplan in public context :
>
> <include>
>   <extension name="public_did">
>     <condition field="destination_number" expression="^(9811806537)$">
>         <action application="transfer" data="5000 XML default"/>
>     </condition>
>   </extension>
> </include>
>
>
>
> Freeswitch logs:
>
> 2016-06-07 05:10:51.000656 [NOTICE] switch_channel.c:1055 New Channel
> sofia/external/00919716517818 at 192.168.20.36:5060
> [33e4e958-2c6e-11e6-9af0-278859dd14e6]
> 2016-06-07 05:10:51.000656 [INFO] mod_dialplan_xml.c:635 Processing
> 00919716517818 <00919716517818>->9811806537 in context public
> 2016-06-07 05:10:51.000656 [NOTICE] switch_ivr.c:1861 Transfer
> sofia/external/00919716517818 at 192.168.20.36:5060 to XML[5000 at default]
> 2016-06-07 05:10:51.000656 [INFO] mod_dialplan_xml.c:635 Processing
> 00919716517818 <00919716517818>->5000 in context default
> 2016-06-07 05:10:51.000656 [NOTICE] sofia_media.c:92 Pre-Answer
> sofia/external/00919716517818 at 192.168.20.36:5060!
> 2016-06-07 05:10:51.000656 [NOTICE] mod_dptools.c:1268 Channel
> [sofia/external/00919716517818 at 192.168.20.36:5060] has been answered
> 2016-06-07 05:10:51.640675 [INFO] switch_rtp.c:5846 Auto Changing port
> from 192.168.20.36:31012 to 103.25.231.101:31012
> 2016-06-07 05:11:06.560680 [NOTICE] sofia.c:952 Hangup sofia/external/
> 00919716517818 at 192.168.20.36:5060 [CS_EXECUTE] [NORMAL_CLEARING]
> 2016-06-07 05:11:06.560680 [NOTICE] switch_core_session.c:1641 Session 262
> (sofia/external/00919716517818 at 192.168.20.36:5060) Ended
> 2016-06-07 05:11:06.560680 [NOTICE] switch_core_session.c:1645 Close
> Channel sofia/external/00919716517818 at 192.168.20.36:5060 [CS_DESTROY]
>
>
> Regards,
> Deepika
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.villasmil.work at gmail.com  Tue Jun  7 20:37:49 2016
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 7 Jun 2016 18:37:49 +0200
Subject: [Freeswitch-users] FS as a SRS
Message-ID: <CAFGRPVqfkVxgfSrUWQXmNWEcSm5X+7_68u=x4qt-pgNXo-PbLw@mail.gmail.com>

Hello,
Is it possible to use FS as a recording server?

We will be receiving INVITEs with a siprec tag and sdp with the
participants. To which FS should answer with 2 rtp ports on which to
receive the audio. We would be receiving 2 rtp streams that we need to
record to a file.

Thanks!

David
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From kathleen at freeswitch.org  Tue Jun  7 21:06:45 2016
From: kathleen at freeswitch.org (Kathleen King)
Date: Tue, 7 Jun 2016 10:06:45 -0700
Subject: [Freeswitch-users] FreeSWITCH training and ClueCon 2016!
Message-ID: <CAOStUPbzsrf+46fKdBVByft7fm400vc=EcEz9z89q0+pU66h-w@mail.gmail.com>

[image: https://cluecon.com] <https://cluecon.com>
[image: https://cluecon.com] <https://cluecon.com>
?
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From zolmed75 at yahoo.com  Tue Jun  7 21:23:50 2016
From: zolmed75 at yahoo.com (Zoltan Medveczky)
Date: Tue, 7 Jun 2016 17:23:50 +0000 (UTC)
Subject: [Freeswitch-users] Slow Memory Leak with FS 1.4.2 ?
References: <1944730894.469007.1465320230315.JavaMail.yahoo.ref@mail.yahoo.com>
Message-ID: <1944730894.469007.1465320230315.JavaMail.yahoo@mail.yahoo.com>

Hi,
I'm running FS 1.4.2 on the following environment:
OS: ?RHEL 6.5CPU: ?x86 64bit quad coreRAM: ?12G

The process has been up for almost 2 months and seems to be consuming an inordinate amount of memory which is slowly increasing over time:
? PID USER ? ? ?PR ?NI ?VIRT ?RES ?SHR S %CPU %MEM ? ?TIME+ ?COMMAND?3404 contactu ?39 ?19 7572m 2.1g 8384 S ?0.7 18.3 578:22.95 freeswitch
600 ? ? ? 8685 ? ? 1 ?0 Apr10 ? ? ? ? ?09:49:55 ./freeswitch -nc -conf /opt/freeswitch-1.4.20/conf -log /var/log/8x8/reach-fs-1.2.0 -db /opt/freeswitch-1.4.20/db

There has been no traffic handled by this process.
Are there any known issues with this version of FS in regards to memory usage?
Thanks
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From carlos.ruizdiaz at gmail.com  Wed Jun  8 04:55:42 2016
From: carlos.ruizdiaz at gmail.com (=?UTF-8?Q?Carlos_Ruiz_D=C3=ADaz?=)
Date: Tue, 7 Jun 2016 19:55:42 -0500
Subject: [Freeswitch-users] ICE problem
Message-ID: <CA+3gBc=1-2H+VNU2ii_7urRRXtBWyPh0K4FxvBCdYKqWRAT_Lg@mail.gmail.com>

Hi guys,

I set up latest FS from master, and there seems to be a problem with the
ICE handshake and establishing audio flow.

The problem that I detected is that ICE setup takes 2 or more seconds to
complete, and around 10% of the time it fails. Same SIPjs configuration
works fine with my current setup, which is based on *git 7cf1238
2015-08-07.*

Is there anything that I need to configure or take into account for latest
master?

Thanks,
-- 
Carlos
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From carlos.ruizdiaz at gmail.com  Wed Jun  8 05:04:06 2016
From: carlos.ruizdiaz at gmail.com (=?UTF-8?Q?Carlos_Ruiz_D=C3=ADaz?=)
Date: Tue, 7 Jun 2016 20:04:06 -0500
Subject: [Freeswitch-users] ICE problem
In-Reply-To: <CA+3gBc=1-2H+VNU2ii_7urRRXtBWyPh0K4FxvBCdYKqWRAT_Lg@mail.gmail.com>
References: <CA+3gBc=1-2H+VNU2ii_7urRRXtBWyPh0K4FxvBCdYKqWRAT_Lg@mail.gmail.com>
Message-ID: <CA+3gBcnNjVjxSpvfjQyOkDjxAqsPQcA_APXDKFbeVDOrE9RzUA@mail.gmail.com>

It looks like ICE is not the problem, but the DTLS handshake. (*)

*2016-06-08 00:57:46.195948 [INFO] switch_core_media.c:6721 Skipping RTCP
ICE (Same as RTP)*
*2016-06-08 00:57:46.195948 [INFO] switch_rtp.c:3328 Activate RTP/RTCP
audio DTLS client*
*2016-06-08 00:57:46.195948 [INFO] switch_rtp.c:3459 Changing audio DTLS
state from OFF to HANDSHAKE*

When it fails, it doesn't go beyond the last line.

I updated openssl to *v1.0.2h*, rebuilt FS, but problem remains.

(*) I occasionally get SIP 488, but that's another story.

Regards,
Carlos

On Tue, Jun 7, 2016 at 7:55 PM, Carlos Ruiz D?az <carlos.ruizdiaz at gmail.com>
wrote:

> Hi guys,
>
> I set up latest FS from master, and there seems to be a problem with the
> ICE handshake and establishing audio flow.
>
> The problem that I detected is that ICE setup takes 2 or more seconds to
> complete, and around 10% of the time it fails. Same SIPjs configuration
> works fine with my current setup, which is based on *git 7cf1238
> 2015-08-07.*
>
> Is there anything that I need to configure or take into account for latest
> master?
>
> Thanks,
> --
> Carlos
>



-- 
Carlos
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From deepikay at iiitd.ac.in  Wed Jun  8 08:35:45 2016
From: deepikay at iiitd.ac.in (Deepika Yadav)
Date: Wed, 8 Jun 2016 10:05:45 +0530
Subject: [Freeswitch-users] GSM gateway to Freeswitch audio problem
In-Reply-To: <CALXCt0p5+GhCJycbSrx0+AA6-PL4tzjcUjvcbcowimak3ndW1Q@mail.gmail.com>
References: <CAM99_Ym407j+aXehJDgYkeTpgX7UDZHn4sdt_SXEi87bkHzatg@mail.gmail.com>
	<CALXCt0p5+GhCJycbSrx0+AA6-PL4tzjcUjvcbcowimak3ndW1Q@mail.gmail.com>
Message-ID: <CAM99_Y=hmf06LqOU9dRzVLqGz9ijRmWJpbAVtfmMtHP0vST5Cw@mail.gmail.com>

Yeah, I removed GSM still no audio but only noise

Codecs PCMU and PCMA seems to be matched

Here are the logs :

http://pastebin.com/VJdjWPGD

On Tue, Jun 7, 2016 at 9:13 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> Delete gsm from.codecs (both inbound and outbound) and try again.
>
> Also, activate debug on console, so you can see more of what happens.
>
> Read the troubleshooting and debugging techniques from
> www.freeswitch.org/confluence
>
> sent from mobile
> cell: +39 347 266 56 18
> Giovanni Maruzzelli
> OpenTelecom.IT
> Il 07/Giu/2016 17:32, "Deepika Yadav" <deepikay at iiitd.ac.in> ha scritto:
>
>> Hi,
>>
>> I want to make calls forwarded from GSM gateway to Freeswitch extension
>>
>> When I dial a 10- digit mobile number of a sim configured in the GSM
>> gateway in a way that it forwards the incoming call to freeswitch
>> extension. I do not hear anything except for some background noise
>>
>> Freeswitch codec configuration :
>>
>>  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA"/>
>> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=GSM,PCMU,PCMA"/>
>>
>> GSM- Gateway Codec settings :
>>
>> priority 1 - G.711 u-law
>> priority 2 - G.711 a-law
>>
>> Dialplan in public context :
>>
>> <include>
>>   <extension name="public_did">
>>     <condition field="destination_number" expression="^(9811806537)$">
>>         <action application="transfer" data="5000 XML default"/>
>>     </condition>
>>   </extension>
>> </include>
>>
>>
>>
>> Freeswitch logs:
>>
>> 2016-06-07 05:10:51.000656 [NOTICE] switch_channel.c:1055 New Channel
>> sofia/external/00919716517818 at 192.168.20.36:5060
>> [33e4e958-2c6e-11e6-9af0-278859dd14e6]
>> 2016-06-07 05:10:51.000656 [INFO] mod_dialplan_xml.c:635 Processing
>> 00919716517818 <00919716517818>->9811806537 in context public
>> 2016-06-07 05:10:51.000656 [NOTICE] switch_ivr.c:1861 Transfer
>> sofia/external/00919716517818 at 192.168.20.36:5060 to XML[5000 at default]
>> 2016-06-07 05:10:51.000656 [INFO] mod_dialplan_xml.c:635 Processing
>> 00919716517818 <00919716517818>->5000 in context default
>> 2016-06-07 05:10:51.000656 [NOTICE] sofia_media.c:92 Pre-Answer
>> sofia/external/00919716517818 at 192.168.20.36:5060!
>> 2016-06-07 05:10:51.000656 [NOTICE] mod_dptools.c:1268 Channel
>> [sofia/external/00919716517818 at 192.168.20.36:5060] has been answered
>> 2016-06-07 05:10:51.640675 [INFO] switch_rtp.c:5846 Auto Changing port
>> from 192.168.20.36:31012 to 103.25.231.101:31012
>> 2016-06-07 05:11:06.560680 [NOTICE] sofia.c:952 Hangup sofia/external/
>> 00919716517818 at 192.168.20.36:5060 [CS_EXECUTE] [NORMAL_CLEARING]
>> 2016-06-07 05:11:06.560680 [NOTICE] switch_core_session.c:1641 Session
>> 262 (sofia/external/00919716517818 at 192.168.20.36:5060) Ended
>> 2016-06-07 05:11:06.560680 [NOTICE] switch_core_session.c:1645 Close
>> Channel sofia/external/00919716517818 at 192.168.20.36:5060 [CS_DESTROY]
>>
>>
>> Regards,
>> Deepika
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kamil.nigmatullin at gmail.com  Wed Jun  8 10:55:40 2016
From: kamil.nigmatullin at gmail.com (Kamil Nigmatullin)
Date: Wed, 8 Jun 2016 12:55:40 +0600
Subject: [Freeswitch-users] Hanhup hook problem when I make a-leg only call
Message-ID: <CA+Xtief0zT9Pd7WgxuwTaheE6KQyh2KH1jpRBaBSkvhfoArWYg@mail.gmail.com>

Hello,

I want to make a program, that makes a call to the user and calls my script
when it ends.

I run the following script:

dialA =
"{origination_caller_id_number="..telephone_from..",ignore_early_media=true}sofia/gateway/iskratel/"..telephone_to


while(legA:ready() and dispoA ~= "ANSWER") do
  dispoA = legA:getVariable("endpoint_disposition")
  freeswitch.consoleLog("INFO","Leg A disposition is '" .. dispoA .. "'\n")
  legA:execute("set", "api_hangup_hook=luarun autoinform_cleanup.lua");
  legA:execute("set", "session_in_hangup_hook=true");
  legA:execute("playback", "/usr/local/freeswitch/sounds/alert.wav");
  legA:execute("hangup");
end

and simple log of autoinform_cleanup.lua

dat = env:serialize()
freeswitch.consoleLog("INFO","Here's everything:\n" .. dat .. "\n")


It makes a call, but when autoinform_cleanup.lua is called
I see this in console

/usr/local/freeswitch/scripts/autoinform_cleanup.lua:1: attempt to index
global 'env' (a nil value)


-- 
Kamil Nigmatullin
Tel: 77272323748
mob: 7 (707) 2517003
Skype: kamil.nigmatullin
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From benjamin.cropley at gmail.com  Wed Jun  8 11:30:40 2016
From: benjamin.cropley at gmail.com (Benjamin Cropley)
Date: Wed, 8 Jun 2016 08:30:40 +0100
Subject: [Freeswitch-users] GSM gateway to Freeswitch audio problem
In-Reply-To: <CAM99_Y=hmf06LqOU9dRzVLqGz9ijRmWJpbAVtfmMtHP0vST5Cw@mail.gmail.com>
References: <CAM99_Ym407j+aXehJDgYkeTpgX7UDZHn4sdt_SXEi87bkHzatg@mail.gmail.com>
	<CALXCt0p5+GhCJycbSrx0+AA6-PL4tzjcUjvcbcowimak3ndW1Q@mail.gmail.com>
	<CAM99_Y=hmf06LqOU9dRzVLqGz9ijRmWJpbAVtfmMtHP0vST5Cw@mail.gmail.com>
Message-ID: <CAF7jizKzvPKQu+pDSLAiYdRT0NWs-R6+TFtAc+F65TiVJ5NjBw@mail.gmail.com>

Deepika,

Have you tried doing packet capture and looking at the RTP data to be sure
audio is 100% being transmitted between the two endpoints?

By that I mean..

- Do a packet capture on FreeSWITCH server and check RTP is sent and
received
- Do the same thing, but from the GSM gateway (if possible) and check the
same.

p.s. your log file is truncated on the longer lines :(

Ben

On Wed, Jun 8, 2016 at 5:35 AM, Deepika Yadav <deepikay at iiitd.ac.in> wrote:

> Yeah, I removed GSM still no audio but only noise
>
> Codecs PCMU and PCMA seems to be matched
>
> Here are the logs :
>
> http://pastebin.com/VJdjWPGD
>
> On Tue, Jun 7, 2016 at 9:13 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> Delete gsm from.codecs (both inbound and outbound) and try again.
>>
>> Also, activate debug on console, so you can see more of what happens.
>>
>> Read the troubleshooting and debugging techniques from
>> www.freeswitch.org/confluence
>>
>> sent from mobile
>> cell: +39 347 266 56 18
>> Giovanni Maruzzelli
>> OpenTelecom.IT
>> Il 07/Giu/2016 17:32, "Deepika Yadav" <deepikay at iiitd.ac.in> ha scritto:
>>
>>> Hi,
>>>
>>> I want to make calls forwarded from GSM gateway to Freeswitch extension
>>>
>>> When I dial a 10- digit mobile number of a sim configured in the GSM
>>> gateway in a way that it forwards the incoming call to freeswitch
>>> extension. I do not hear anything except for some background noise
>>>
>>> Freeswitch codec configuration :
>>>
>>>  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA"/>
>>> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=GSM,PCMU,PCMA"/>
>>>
>>> GSM- Gateway Codec settings :
>>>
>>> priority 1 - G.711 u-law
>>> priority 2 - G.711 a-law
>>>
>>> Dialplan in public context :
>>>
>>> <include>
>>>   <extension name="public_did">
>>>     <condition field="destination_number" expression="^(9811806537)$">
>>>         <action application="transfer" data="5000 XML default"/>
>>>     </condition>
>>>   </extension>
>>> </include>
>>>
>>>
>>>
>>> Freeswitch logs:
>>>
>>> 2016-06-07 05:10:51.000656 [NOTICE] switch_channel.c:1055 New Channel
>>> sofia/external/00919716517818 at 192.168.20.36:5060
>>> [33e4e958-2c6e-11e6-9af0-278859dd14e6]
>>> 2016-06-07 05:10:51.000656 [INFO] mod_dialplan_xml.c:635 Processing
>>> 00919716517818 <00919716517818>->9811806537 in context public
>>> 2016-06-07 05:10:51.000656 [NOTICE] switch_ivr.c:1861 Transfer
>>> sofia/external/00919716517818 at 192.168.20.36:5060 to XML[5000 at default]
>>> 2016-06-07 05:10:51.000656 [INFO] mod_dialplan_xml.c:635 Processing
>>> 00919716517818 <00919716517818>->5000 in context default
>>> 2016-06-07 05:10:51.000656 [NOTICE] sofia_media.c:92 Pre-Answer
>>> sofia/external/00919716517818 at 192.168.20.36:5060!
>>> 2016-06-07 05:10:51.000656 [NOTICE] mod_dptools.c:1268 Channel
>>> [sofia/external/00919716517818 at 192.168.20.36:5060] has been answered
>>> 2016-06-07 05:10:51.640675 [INFO] switch_rtp.c:5846 Auto Changing port
>>> from 192.168.20.36:31012 to 103.25.231.101:31012
>>> 2016-06-07 05:11:06.560680 [NOTICE] sofia.c:952 Hangup sofia/external/
>>> 00919716517818 at 192.168.20.36:5060 [CS_EXECUTE] [NORMAL_CLEARING]
>>> 2016-06-07 05:11:06.560680 [NOTICE] switch_core_session.c:1641 Session
>>> 262 (sofia/external/00919716517818 at 192.168.20.36:5060) Ended
>>> 2016-06-07 05:11:06.560680 [NOTICE] switch_core_session.c:1645 Close
>>> Channel sofia/external/00919716517818 at 192.168.20.36:5060 [CS_DESTROY]
>>>
>>>
>>> Regards,
>>> Deepika
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bipin at xbipin.com  Wed Jun  8 12:15:12 2016
From: bipin at xbipin.com (Bipin Patel)
Date: Wed, 8 Jun 2016 12:15:12 +0400
Subject: [Freeswitch-users] GSM gateway to Freeswitch audio problem
In-Reply-To: <CAF7jizKzvPKQu+pDSLAiYdRT0NWs-R6+TFtAc+F65TiVJ5NjBw@mail.gmail.com>
References: <CAM99_Ym407j+aXehJDgYkeTpgX7UDZHn4sdt_SXEi87bkHzatg@mail.gmail.com>
	<CALXCt0p5+GhCJycbSrx0+AA6-PL4tzjcUjvcbcowimak3ndW1Q@mail.gmail.com>
	<CAM99_Y=hmf06LqOU9dRzVLqGz9ijRmWJpbAVtfmMtHP0vST5Cw@mail.gmail.com>
	<CAF7jizKzvPKQu+pDSLAiYdRT0NWs-R6+TFtAc+F65TiVJ5NjBw@mail.gmail.com>
Message-ID: <cdce67a9-17c0-d2fe-1e1e-684604f74bd9@xbipin.com>

do this in ur bridge statement in the dialplan, repalce $1 with 
destination number and gsm01 with ur gsm device name

<action application="bridge" 
data="{absolute_codec_string=PCMU}gsmopen/gsm01/$1"/>

Regards,
Bipin


------------------------------------------------------------------------
-------- Original Message --------
Subject: Re: [Freeswitch-users] GSM gateway to Freeswitch audio problem
From: Benjamin Cropley <benjamin.cropley at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date: 6/8/2016, 11:30:40 AM
> Deepika,
>
> Have you tried doing packet capture and looking at the RTP data to be 
> sure audio is 100% being transmitted between the two endpoints?
>
> By that I mean..
>
> - Do a packet capture on FreeSWITCH server and check RTP is sent and 
> received
> - Do the same thing, but from the GSM gateway (if possible) and check 
> the same.
>
> p.s. your log file is truncated on the longer lines :(
>
> Ben
>
> On Wed, Jun 8, 2016 at 5:35 AM, Deepika Yadav <deepikay at iiitd.ac.in 
> <mailto:deepikay at iiitd.ac.in>> wrote:
>
>     Yeah, I removed GSM still no audio but only noise
>
>     Codecs PCMU and PCMA seems to be matched
>
>     Here are the logs :
>
>     http://pastebin.com/VJdjWPGD
>
>     On Tue, Jun 7, 2016 at 9:13 PM, Giovanni Maruzzelli
>     <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>> wrote:
>
>         Delete gsm from.codecs (both inbound and outbound) and try again.
>
>         Also, activate debug on console, so you can see more of what
>         happens.
>
>         Read the troubleshooting and debugging techniques from
>         www.freeswitch.org/confluence
>         <http://www.freeswitch.org/confluence>
>
>         sent from mobile
>         cell: +39 347 266 56 18
>         Giovanni Maruzzelli
>         OpenTelecom.IT
>
>         Il 07/Giu/2016 17:32, "Deepika Yadav" <deepikay at iiitd.ac.in
>         <mailto:deepikay at iiitd.ac.in>> ha scritto:
>
>             Hi,
>
>             I want to make calls forwarded from GSM gateway to
>             Freeswitch extension
>
>             When I dial a 10- digit mobile number of a sim configured
>             in the GSM gateway in a way that it forwards the incoming
>             call to freeswitch extension. I do not hear anything
>             except for some background noise
>
>             Freeswitch codec configuration :
>
>              <X-PRE-PROCESS cmd="set"
>             data="global_codec_prefs=GSM,PCMU,PCMA"/>
>             <X-PRE-PROCESS cmd="set"
>             data="outbound_codec_prefs=GSM,PCMU,PCMA"/>
>
>             GSM- Gateway Codec settings :
>
>             priority 1 - G.711 u-law
>             priority 2 - G.711 a-law
>
>             Dialplan in public context :
>
>             <include>
>               <extension name="public_did">
>                 <condition field="destination_number"
>             expression="^(9811806537)$">
>                     <action application="transfer" data="5000 XML
>             default"/>
>                 </condition>
>               </extension>
>             </include>
>
>
>
>             Freeswitch logs:
>
>             2016-06-07 05:10:51.000656 [NOTICE] switch_channel.c:1055
>             New Channel
>             sofia/external/00919716517818 at 192.168.20.36:5060
>             <http://00919716517818 at 192.168.20.36:5060>
>             [33e4e958-2c6e-11e6-9af0-278859dd14e6]
>             2016-06-07 05:10:51.000656 [INFO] mod_dialplan_xml.c:635
>             Processing 00919716517818 <00919716517818>->9811806537 in
>             context public
>             2016-06-07 05:10:51.000656 [NOTICE] switch_ivr.c:1861
>             Transfer sofia/external/00919716517818 at 192.168.20.36:5060
>             <http://00919716517818 at 192.168.20.36:5060> to
>             XML[5000 at default]
>             2016-06-07 05:10:51.000656 [INFO] mod_dialplan_xml.c:635
>             Processing 00919716517818 <00919716517818>->5000 in
>             context default
>             2016-06-07 05:10:51.000656 [NOTICE] sofia_media.c:92
>             Pre-Answer
>             sofia/external/00919716517818 at 192.168.20.36:5060
>             <http://00919716517818 at 192.168.20.36:5060>!
>             2016-06-07 05:10:51.000656 [NOTICE] mod_dptools.c:1268
>             Channel [sofia/external/00919716517818 at 192.168.20.36:5060
>             <http://00919716517818 at 192.168.20.36:5060>] has been answered
>             2016-06-07 05:10:51.640675 [INFO] switch_rtp.c:5846 Auto
>             Changing port from 192.168.20.36:31012
>             <http://192.168.20.36:31012> to 103.25.231.101:31012
>             <http://103.25.231.101:31012>
>             2016-06-07 05:11:06.560680 [NOTICE] sofia.c:952 Hangup
>             sofia/external/00919716517818 at 192.168.20.36:5060
>             <http://00919716517818 at 192.168.20.36:5060> [CS_EXECUTE]
>             [NORMAL_CLEARING]
>             2016-06-07 05:11:06.560680 [NOTICE]
>             switch_core_session.c:1641 Session 262
>             (sofia/external/00919716517818 at 192.168.20.36:5060
>             <http://00919716517818 at 192.168.20.36:5060>) Ended
>             2016-06-07 05:11:06.560680 [NOTICE]
>             switch_core_session.c:1645 Close Channel
>             sofia/external/00919716517818 at 192.168.20.36:5060
>             <http://00919716517818 at 192.168.20.36:5060> [CS_DESTROY]
>
>
>             Regards,
>             Deepika
>
>
>             _________________________________________________________________________
>             Professional FreeSWITCH Consulting Services:
>             consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>             http://www.freeswitchsolutions.com
>
>             Official FreeSWITCH Sites
>             http://www.freeswitch.org
>             http://confluence.freeswitch.org
>             http://www.cluecon.com
>
>             FreeSWITCH-users mailing list
>             FreeSWITCH-users at lists.freeswitch.org
>             <mailto:FreeSWITCH-users at lists.freeswitch.org>
>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>             http://www.freeswitch.org
>
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         http://www.freeswitchsolutions.com
>
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://confluence.freeswitch.org
>         http://www.cluecon.com
>
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From miguelaustro at gmail.com  Wed Jun  8 14:40:00 2016
From: miguelaustro at gmail.com (Miguel Oyarzo)
Date: Wed, 8 Jun 2016 20:40:00 +1000
Subject: [Freeswitch-users] FS as a SRS
In-Reply-To: <CAFGRPVqfkVxgfSrUWQXmNWEcSm5X+7_68u=x4qt-pgNXo-PbLw@mail.gmail.com>
References: <CAFGRPVqfkVxgfSrUWQXmNWEcSm5X+7_68u=x4qt-pgNXo-PbLw@mail.gmail.com>
Message-ID: <CA+d2mV-5pG-XCy1w=sCNk5gMduy6row1YYM+7zS7dkfomqyq2w@mail.gmail.com>

Nighter FS nor Asterisk will process two same-type RTP streams in the same
SDP. The 2nd RTP streams will be declined and answered with a port 0
(200OK), as suggested by rfc3264.

Regards,

On Wed, Jun 8, 2016 at 2:37 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello,
> Is it possible to use FS as a recording server?
>
> We will be receiving INVITEs with a siprec tag and sdp with the
> participants. To which FS should answer with 2 rtp ports on which to
> receive the audio. We would be receiving 2 rtp streams that we need to
> record to a file.
>
> Thanks!
>
> David
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
"Do not use a cannon to kill a mosquito" (Confucius)

Miguel Oyarzo
VoIP Engineer && DevOps
Linux User: # 483188 - counter.li.org
http://au.linkedin.com/in/mikeaustralia
Melbourne, Australia
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From david.villasmil.work at gmail.com  Wed Jun  8 16:03:08 2016
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 8 Jun 2016 14:03:08 +0200
Subject: [Freeswitch-users] FS as a SRS
In-Reply-To: <CA+d2mV-5pG-XCy1w=sCNk5gMduy6row1YYM+7zS7dkfomqyq2w@mail.gmail.com>
References: <CAFGRPVqfkVxgfSrUWQXmNWEcSm5X+7_68u=x4qt-pgNXo-PbLw@mail.gmail.com>
	<CA+d2mV-5pG-XCy1w=sCNk5gMduy6row1YYM+7zS7dkfomqyq2w@mail.gmail.com>
Message-ID: <CAFGRPVoGxQvn=PSK1Miq5QkpjrZWROoGmXSzf+XMh6titnct3A@mail.gmail.com>

Hello Miguel,

Thanks for your reply, but I think maybe I didn't explain myself correctly,
let me try again:

A recording server receives INVITEs including multiple "m=" which will be
used as rtp source so the server can record an ongoing conversation
happening somewhere else. As per RFC4566 an SDP may have as many m= as it
wants (https://tools.ietf.org/html/rfc4566#page-22).

The idea is to use SIPREC the following way:

The voice server will choose the call to record and send a 3rd leg to the
recording server, which will receive the INVITE ans answer with 200 OK and
start recording the rtps.

Regards,

David

On Wed, Jun 8, 2016 at 12:40 PM, Miguel Oyarzo <miguelaustro at gmail.com>
wrote:

>
>
> Nighter FS nor Asterisk will process two same-type RTP streams in the same
> SDP. The 2nd RTP streams will be declined and answered with a port 0
> (200OK), as suggested by rfc3264.
>
> Regards,
>
> On Wed, Jun 8, 2016 at 2:37 AM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Hello,
>> Is it possible to use FS as a recording server?
>>
>> We will be receiving INVITEs with a siprec tag and sdp with the
>> participants. To which FS should answer with 2 rtp ports on which to
>> receive the audio. We would be receiving 2 rtp streams that we need to
>> record to a file.
>>
>> Thanks!
>>
>> David
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> "Do not use a cannon to kill a mosquito" (Confucius)
>
> Miguel Oyarzo
> VoIP Engineer && DevOps
> Linux User: # 483188 - counter.li.org
> http://au.linkedin.com/in/mikeaustralia
> Melbourne, Australia
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From krice at freeswitch.org  Wed Jun  8 16:34:28 2016
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 8 Jun 2016 07:34:28 -0500
Subject: [Freeswitch-users] FS as a SRS
In-Reply-To: <CAFGRPVoGxQvn=PSK1Miq5QkpjrZWROoGmXSzf+XMh6titnct3A@mail.gmail.com>
References: <CAFGRPVqfkVxgfSrUWQXmNWEcSm5X+7_68u=x4qt-pgNXo-PbLw@mail.gmail.com>
	<CA+d2mV-5pG-XCy1w=sCNk5gMduy6row1YYM+7zS7dkfomqyq2w@mail.gmail.com>
	<CAFGRPVoGxQvn=PSK1Miq5QkpjrZWROoGmXSzf+XMh6titnct3A@mail.gmail.com>
Message-ID: <27303E24-E1BB-4603-BF4B-3B77D870C92D@freeswitch.org>

This is not currently implemented in FreeSWITCH. It may be worth contacting consulting at freeswitch.org and talking with us about implementing this

Sent from my iPhone

> On Jun 8, 2016, at 7:03 AM, David Villasmil <david.villasmil.work at gmail.com> wrote:
> 
> Hello Miguel,
> 
> Thanks for your reply, but I think maybe I didn't explain myself correctly, let me try again:
> 
> A recording server receives INVITEs including multiple "m=" which will be used as rtp source so the server can record an ongoing conversation happening somewhere else. As per RFC4566 an SDP may have as many m= as it wants (https://tools.ietf.org/html/rfc4566#page-22).
> 
> The idea is to use SIPREC the following way:
> 
> The voice server will choose the call to record and send a 3rd leg to the recording server, which will receive the INVITE ans answer with 200 OK and start recording the rtps.
> 
> Regards,
> 
> David
> 
>> On Wed, Jun 8, 2016 at 12:40 PM, Miguel Oyarzo <miguelaustro at gmail.com> wrote:
>> 
>> 
>> Nighter FS nor Asterisk will process two same-type RTP streams in the same SDP. The 2nd RTP streams will be declined and answered with a port 0 (200OK), as suggested by rfc3264.
>> 
>> Regards,
>> 
>>> On Wed, Jun 8, 2016 at 2:37 AM, David Villasmil <david.villasmil.work at gmail.com> wrote:
>>> Hello,
>>> Is it possible to use FS as a recording server?
>>> 
>>> We will be receiving INVITEs with a siprec tag and sdp with the participants. To which FS should answer with 2 rtp ports on which to receive the audio. We would be receiving 2 rtp streams that we need to record to a file.
>>> 
>>> Thanks!
>>> 
>>> David
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> 
>> -- 
>> "Do not use a cannon to kill a mosquito" (Confucius)
>> 
>> Miguel Oyarzo
>> VoIP Engineer && DevOps 
>> Linux User: # 483188 - counter.li.org
>> http://au.linkedin.com/in/mikeaustralia 
>> Melbourne, Australia
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From steveayre at gmail.com  Wed Jun  8 16:38:46 2016
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 8 Jun 2016 13:38:46 +0100
Subject: [Freeswitch-users] Slow Memory Leak with FS 1.4.2 ?
In-Reply-To: <1944730894.469007.1465320230315.JavaMail.yahoo@mail.yahoo.com>
References: <1944730894.469007.1465320230315.JavaMail.yahoo.ref@mail.yahoo.com>
	<1944730894.469007.1465320230315.JavaMail.yahoo@mail.yahoo.com>
Message-ID: <CAFiqYukf26poFXVp6cpjtk5Qh5za7smyweWHPLgwdCkzJDmeww@mail.gmail.com>

That version is very out of date. Try upgrading to at least 1.4.26.

On 7 June 2016 at 18:23, Zoltan Medveczky <zolmed75 at yahoo.com> wrote:

> Hi,
>
> I'm running FS 1.4.2 on the following environment:
>
> OS:  RHEL 6.5
> CPU:  x86 64bit quad core
> RAM:  12G
>
> The process has been up for almost 2 months and seems to be consuming an
> inordinate amount of memory which is slowly increasing over time:
>
>   PID USER      PR  NI  VIRT  *RES*  SHR S %CPU %MEM    TIME+  COMMAND
>  3404 contactu  39  19 7572m *2.1g* 8384 S  0.7 18.3 578:22.95 freeswitch
>
> 600       8685     1  0 *Apr10* ?        09:49:55 ./freeswitch -nc -conf
> /opt/freeswitch-1.4.20/conf -log /var/log/8x8/reach-fs-1.2.0 -db
> /opt/freeswitch-1.4.20/db
>
> There has been no traffic handled by this process.
>
> Are there any known issues with this version of FS in regards to memory
> usage?
>
> Thanks
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From me at nevian.org  Wed Jun  8 17:21:27 2016
From: me at nevian.org (Serge S. Yuriev)
Date: Wed, 8 Jun 2016 16:21:27 +0300
Subject: [Freeswitch-users] mod_H.323, working stable ?
In-Reply-To: <VI1PR06MB131106332C9A7751D6917B16EF5D0@VI1PR06MB1311.eurprd06.prod.outlook.com>
References: <VI1PR06MB131106332C9A7751D6917B16EF5D0@VI1PR06MB1311.eurprd06.prod.outlook.com>
Message-ID: <57581BD7.6060205@nevian.org>

Hi,

About five or six years ago I have to use YATE as proto converter and 
after some time (patched) mod_h323 in semi-prod (10-15k calls/day).
mod_OPAL was absolutely broken and outdated till 2 or 3 years ago it got 
major update and IIRC even worked (as of fall'15/winter'16) but recent 
tests few days ago completely failed.

Fairly speaking we have one more: mod_ooh323 - based on same lib from 
Objective Systems as Asterisk one. It lives in separate branch.
As of today simple test call worked w/small patch :)

All of underlying libs seems to be abandoned.

On 07/06/16 11:44, Marcel Haldemann wrote:
> Howdy fellow Freeswich-Users,
>
> We are using FreeSWITCH since while now with SIP. However, recently I
> got ask what I would take to support H.323.
>
> I then checked the Docs and the commits in the git repos for mod_h323
> and mod_opal.
>
> The changes in the repos seems to be similar (not mutch (only some
> VS2015 updates) going on in recent years). Regarding the Docs it seems
> better to use mod_h323 rather than  mod_opal, as it is in a ?beta? stage
> (info from 2008, maybe outdated ?).
>
> The underlying project H323plus seems to have stopped development in 2014.
>
> Is there someone, currently using H.323 with FreeSwitch and made it
> working stable (preferably using FS 1.6.x) ?
>
> If so, any hints on what mod to use (mod_h323 or mod_opal) ? any traps ?
> regarding docs in the H323 mod the trunk version of PTLIB does not work.
> What Version did u use ?
>
> I think it would be better to use mod_h323 or is it suggested to use
> mod_opal ?
>
> What are your experiences in H.323 with FreeSwitch, does it work stable ?
>
> Trying to get an impression whether it?s a good idea to use the H.323
> mod (or opal), I?m happy about any feedback. Also if someone countered
> unresolvable problems with the H.323 mod.

-- 
Serge S. Yuriev
Lead VoIP engineer



From eduardo at ctidealer.com.br  Wed Jun  8 18:27:54 2016
From: eduardo at ctidealer.com.br (edrperes)
Date: Wed, 8 Jun 2016 07:27:54 -0700 (MST)
Subject: [Freeswitch-users] GsmOpen call control
Message-ID: <1465396074152-7596221.post@n2.nabble.com>

I am testing GsmOpen on FreeSwitch 1.7.0 64 bits on Windows 10

The call is made, the target phone rings and the call is answered.

But no "answered" event arrives...

Then only one event that arrive from that call is "early"

Is it normal? 

When I make a call to a soffone through sofia SIP many events happen ring,
answered, hangup... 

Why not through GsmOpen?



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/GsmOpen-call-control-tp7596221.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From gmaruzz at gmail.com  Wed Jun  8 19:30:19 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 8 Jun 2016 17:30:19 +0200
Subject: [Freeswitch-users] GsmOpen call control
In-Reply-To: <1465396074152-7596221.post@n2.nabble.com>
References: <1465396074152-7596221.post@n2.nabble.com>
Message-ID: <CALXCt0rGxEyu2A6drM1t7cFhUhEcLRHG57eg10jpy5LYEo-ieg@mail.gmail.com>

Probably your dongle is not a supported model, or you have not unblocked it
for voice...



On Wed, Jun 8, 2016 at 4:27 PM, edrperes <eduardo at ctidealer.com.br> wrote:

> I am testing GsmOpen on FreeSwitch 1.7.0 64 bits on Windows 10
>
> The call is made, the target phone rings and the call is answered.
>
> But no "answered" event arrives...
>
> Then only one event that arrive from that call is "early"
>
> Is it normal?
>
> When I make a call to a soffone through sofia SIP many events happen ring,
> answered, hangup...
>
> Why not through GsmOpen?
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/GsmOpen-call-control-tp7596221.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From deepikay at iiitd.ac.in  Wed Jun  8 19:41:00 2016
From: deepikay at iiitd.ac.in (Deepika Yadav)
Date: Wed, 8 Jun 2016 21:11:00 +0530
Subject: [Freeswitch-users] GSM gateway to Freeswitch audio problem
In-Reply-To: <CAF7jizKzvPKQu+pDSLAiYdRT0NWs-R6+TFtAc+F65TiVJ5NjBw@mail.gmail.com>
References: <CAM99_Ym407j+aXehJDgYkeTpgX7UDZHn4sdt_SXEi87bkHzatg@mail.gmail.com>
	<CALXCt0p5+GhCJycbSrx0+AA6-PL4tzjcUjvcbcowimak3ndW1Q@mail.gmail.com>
	<CAM99_Y=hmf06LqOU9dRzVLqGz9ijRmWJpbAVtfmMtHP0vST5Cw@mail.gmail.com>
	<CAF7jizKzvPKQu+pDSLAiYdRT0NWs-R6+TFtAc+F65TiVJ5NjBw@mail.gmail.com>
Message-ID: <CAM99_YnwVGYO+JOv__Wwe4Uhx8tjTJZWmpCHnfFEn6oOBJp4ZA@mail.gmail.com>

I have captured the packets at both the endpoints:

Freeswitch does not receive RTP packets, logs:

http://pasted.co/a1a50e4e

where, when I make call from a registered SIP account configured at a
softphone, I clearly hear sound form ivr-demo sound.


GSM VOIP Gateway is sending RTP packets:

http://pastebin.com/0xmBB8rh


Codecs settings at both the ends is PCMA and PCMU only

Thanks,
Deepika


<http://pastebin.com/0xmBB8rh>


On Wed, Jun 8, 2016 at 1:00 PM, Benjamin Cropley <benjamin.cropley at gmail.com
> wrote:

> Deepika,
>
> Have you tried doing packet capture and looking at the RTP data to be sure
> audio is 100% being transmitted between the two endpoints?
>
> By that I mean..
>
> - Do a packet capture on FreeSWITCH server and check RTP is sent and
> received
> - Do the same thing, but from the GSM gateway (if possible) and check the
> same.
>
> p.s. your log file is truncated on the longer lines :(
>
> Ben
>
> On Wed, Jun 8, 2016 at 5:35 AM, Deepika Yadav <deepikay at iiitd.ac.in>
> wrote:
>
>> Yeah, I removed GSM still no audio but only noise
>>
>> Codecs PCMU and PCMA seems to be matched
>>
>> Here are the logs :
>>
>> http://pastebin.com/VJdjWPGD
>>
>> On Tue, Jun 7, 2016 at 9:13 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>> wrote:
>>
>>> Delete gsm from.codecs (both inbound and outbound) and try again.
>>>
>>> Also, activate debug on console, so you can see more of what happens.
>>>
>>> Read the troubleshooting and debugging techniques from
>>> www.freeswitch.org/confluence
>>>
>>> sent from mobile
>>> cell: +39 347 266 56 18
>>> Giovanni Maruzzelli
>>> OpenTelecom.IT
>>> Il 07/Giu/2016 17:32, "Deepika Yadav" <deepikay at iiitd.ac.in> ha scritto:
>>>
>>>> Hi,
>>>>
>>>> I want to make calls forwarded from GSM gateway to Freeswitch extension
>>>>
>>>> When I dial a 10- digit mobile number of a sim configured in the GSM
>>>> gateway in a way that it forwards the incoming call to freeswitch
>>>> extension. I do not hear anything except for some background noise
>>>>
>>>> Freeswitch codec configuration :
>>>>
>>>>  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA"/>
>>>> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=GSM,PCMU,PCMA"/>
>>>>
>>>> GSM- Gateway Codec settings :
>>>>
>>>> priority 1 - G.711 u-law
>>>> priority 2 - G.711 a-law
>>>>
>>>> Dialplan in public context :
>>>>
>>>> <include>
>>>>   <extension name="public_did">
>>>>     <condition field="destination_number" expression="^(9811806537)$">
>>>>         <action application="transfer" data="5000 XML default"/>
>>>>     </condition>
>>>>   </extension>
>>>> </include>
>>>>
>>>>
>>>>
>>>> Freeswitch logs:
>>>>
>>>> 2016-06-07 05:10:51.000656 [NOTICE] switch_channel.c:1055 New Channel
>>>> sofia/external/00919716517818 at 192.168.20.36:5060
>>>> [33e4e958-2c6e-11e6-9af0-278859dd14e6]
>>>> 2016-06-07 05:10:51.000656 [INFO] mod_dialplan_xml.c:635 Processing
>>>> 00919716517818 <00919716517818>->9811806537 in context public
>>>> 2016-06-07 05:10:51.000656 [NOTICE] switch_ivr.c:1861 Transfer
>>>> sofia/external/00919716517818 at 192.168.20.36:5060 to XML[5000 at default]
>>>> 2016-06-07 05:10:51.000656 [INFO] mod_dialplan_xml.c:635 Processing
>>>> 00919716517818 <00919716517818>->5000 in context default
>>>> 2016-06-07 05:10:51.000656 [NOTICE] sofia_media.c:92 Pre-Answer
>>>> sofia/external/00919716517818 at 192.168.20.36:5060!
>>>> 2016-06-07 05:10:51.000656 [NOTICE] mod_dptools.c:1268 Channel
>>>> [sofia/external/00919716517818 at 192.168.20.36:5060] has been answered
>>>> 2016-06-07 05:10:51.640675 [INFO] switch_rtp.c:5846 Auto Changing port
>>>> from 192.168.20.36:31012 to 103.25.231.101:31012
>>>> 2016-06-07 05:11:06.560680 [NOTICE] sofia.c:952 Hangup sofia/external/
>>>> 00919716517818 at 192.168.20.36:5060 [CS_EXECUTE] [NORMAL_CLEARING]
>>>> 2016-06-07 05:11:06.560680 [NOTICE] switch_core_session.c:1641 Session
>>>> 262 (sofia/external/00919716517818 at 192.168.20.36:5060) Ended
>>>> 2016-06-07 05:11:06.560680 [NOTICE] switch_core_session.c:1645 Close
>>>> Channel sofia/external/00919716517818 at 192.168.20.36:5060 [CS_DESTROY]
>>>>
>>>>
>>>> Regards,
>>>> Deepika
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From eduardo at ctidealer.com.br  Wed Jun  8 19:50:44 2016
From: eduardo at ctidealer.com.br (edrperes)
Date: Wed, 8 Jun 2016 08:50:44 -0700 (MST)
Subject: [Freeswitch-users] GsmOpen call control
In-Reply-To: <1465396074152-7596221.post@n2.nabble.com>
References: <1465396074152-7596221.post@n2.nabble.com>
Message-ID: <1465401044177-7596222.post@n2.nabble.com>

My modem is Huawei e1756 unlocked and voice is enabled

The audio is sending and receiving ok...

The problem is the lack of events like "ringing", "answered", "hangup"...





--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/GsmOpen-call-control-tp7596221p7596222.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From gmaruzz at gmail.com  Wed Jun  8 20:01:08 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 8 Jun 2016 18:01:08 +0200
Subject: [Freeswitch-users] GsmOpen call control
In-Reply-To: <1465401044177-7596222.post@n2.nabble.com>
References: <1465396074152-7596221.post@n2.nabble.com>
	<1465401044177-7596222.post@n2.nabble.com>
Message-ID: <CALXCt0pfG+fJuR8i9zLZ_7qWfUq+PtzLOXJvVcOChjMN=eXbNw@mail.gmail.com>

On Wed, Jun 8, 2016 at 5:50 PM, edrperes <eduardo at ctidealer.com.br> wrote:

> My modem is Huawei e1756 unlocked and voice is enabled
>
> The audio is sending and receiving ok...
>
> The problem is the lack of events like "ringing", "answered", "hangup"...
>
>
>

That's unusual.

Can you follow the instruction on gsmopen page on how to report problems,
and attach all relevant info?

-giovanni



>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/GsmOpen-call-control-tp7596221p7596222.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From lists at kavun.ch  Wed Jun  8 21:22:16 2016
From: lists at kavun.ch (Emrah)
Date: Wed, 8 Jun 2016 19:22:16 +0200
Subject: [Freeswitch-users] Distributed presence
Message-ID: <6F058A02-9808-459C-8C85-AAF1F2C76FFD@kavun.ch>

Hello list,
I have multiple FS servers and I am trying to share presence info across them using a PGSQL database.
I cannot share the same hostname and use PGSQL in the core that way for 2 reasons,
1. The DB is on a remote server and there is some latency. It?s ok to update and show presence but not to conduct all operations. Also, if the DB server fails, everything needs to keep on running normally.
2. I need to be able to distinguish channels and calls on a per server basis.

Worst case, I could probably forfeit point 1 if FS can keep on running if the DB crashes, if it?s the only way. But I cannot share the same hostnames across all boxes.
I already have odbc-dsn configured for my internal profile, as well as:
    <param name="presence-probe-on-register" value="true"/>
    <param name="manage-shared-appearance" value="true"/>
    <param name="dbname" value="sip_presence"/>

For dbname, I put the name of a database I saw FS created and which I assumed would match the purpose? But I couldn?t find much documentation about it. Any hints would be appreciated. That particular table (sip_presence) stays empty and presence isn?t shared.

Thanks for any help and all the best,

Emrah
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From mike at jerris.com  Wed Jun  8 23:05:00 2016
From: mike at jerris.com (Michael Jerris)
Date: Wed, 8 Jun 2016 15:05:00 -0400
Subject: [Freeswitch-users] Slow Memory Leak with FS 1.4.2 ?
In-Reply-To: <CAFiqYukf26poFXVp6cpjtk5Qh5za7smyweWHPLgwdCkzJDmeww@mail.gmail.com>
References: <1944730894.469007.1465320230315.JavaMail.yahoo.ref@mail.yahoo.com>
	<1944730894.469007.1465320230315.JavaMail.yahoo@mail.yahoo.com>
	<CAFiqYukf26poFXVp6cpjtk5Qh5za7smyweWHPLgwdCkzJDmeww@mail.gmail.com>
Message-ID: <933D0A64-8DF6-4658-9159-D54CE9A31F7B@jerris.com>

1.4.26 is way out of date.. Try updating to the latest 1.6 release.


> On Jun 8, 2016, at 8:38 AM, Steven Ayre <steveayre at gmail.com> wrote:
> 
> That version is very out of date. Try upgrading to at least 1.4.26.
> 
> On 7 June 2016 at 18:23, Zoltan Medveczky <zolmed75 at yahoo.com <mailto:zolmed75 at yahoo.com>> wrote:
> Hi,
> 
> I'm running FS 1.4.2 on the following environment:
> 
> OS:  RHEL 6.5
> CPU:  x86 64bit quad core
> RAM:  12G
> 
> The process has been up for almost 2 months and seems to be consuming an inordinate amount of memory which is slowly increasing over time:
> 
>   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+  COMMAND
>  3404 contactu  39  19 7572m 2.1g 8384 S  0.7 18.3 578:22.95 freeswitch
> 
> 600       8685     1  0 Apr10 ?        09:49:55 ./freeswitch -nc -conf /opt/freeswitch-1.4.20/conf -log /var/log/8x8/reach-fs-1.2.0 -db /opt/freeswitch-1.4.20/db
> 
> There has been no traffic handled by this process.
> 
> Are there any known issues with this version of FS in regards to memory usage?
> 
> Thanks
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From matt at supportedbusiness.com  Thu Jun  9 00:21:08 2016
From: matt at supportedbusiness.com (Matt Broad)
Date: Wed, 8 Jun 2016 21:21:08 +0100
Subject: [Freeswitch-users] add modules when installing from the latest
	branch
In-Reply-To: <007501d1c0d2$2febc220$8fc34660$@freeswitch.org>
References: <CAEeV61jv_rX-ebF6O4gM3=DG3jEDdMaBBovYtj_2QAwzozPJrg@mail.gmail.com>
	<CAF7jizJ7yXS5a0QhFm5uY1vZttcmO5HoM3sssCAqLY1NJyO37g@mail.gmail.com>
	<CAEeV61g12Ar2euivZQb+3NkTB7mRO7keRe12AN7kp9J8Do3xZw@mail.gmail.com>
	<007501d1c0d2$2febc220$8fc34660$@freeswitch.org>
Message-ID: <CAEeV61h+5Gy2Wo16jOEze2qPGmrD8q8pb3q01wTViAv2Sz+Z7A@mail.gmail.com>

thanks Ken, just another quick question on this install method; how would I
go about changing the default locations for the scripts?

I would ordinarily use the following: ./configure --with-scriptdir=
/etc/freeswitch/scripts/



thanks
Matt

On 7 June 2016 at 16:35, Ken Rice <krice at freeswitch.org> wrote:

> Easier is probably just apt-cache search freeswitch or apt-cache search
> freeswitch|grep mongo or something like that
>
>
>
> Once you do that you can apt-get install the module you want and then you
> just need to enable it in your
> /etc/freeswitch/autoload_configs/modules.conf.xml
>
>
>
> The entire package system is designed so that you can use
> https://freeswitch.org/stash/projects/FS/repos/freeswitch-config-skel/browse
> to generate your own configuration package with deps on the modules that
> you want to include and just install that and bam you?re good
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Matt Broad
> *Sent:* Tuesday, June 7, 2016 5:50 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] add modules when installing from the
> latest branch
>
>
>
> perfect!
>
>
>
> thanks for the quick response.  Is this listed on the confluence site
> anywhere?  I am happy to add this if given access
>
>
>
> thanks
>
> Matt
>
>
>
> On 7 June 2016 at 11:35, Benjamin Cropley <benjamin.cropley at gmail.com>
> wrote:
>
> Say you wanted to install mod_mongo..
>
>
>
> you can find the packages here
> http://files.freeswitch.org/repo/deb/freeswitch-1.6/dists/jessie/main/binary-amd64/Packages
>
>
>
> apt-get install -y freeswitch-mod-mongo
>
>
>
> then edit the modules conf as you would have done previously
>
>
>
> On Tue, Jun 7, 2016 at 11:20 AM, Matt Broad <matt at supportedbusiness.com>
> wrote:
>
> Hi,
>
>
>
> this may be a really stupid question but here goes:
>
>
>
> I have upgraded a server to debian 8 and have installed freeswitch using
> the latest branch
>
>
> https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie#Debian8Jessie-latestreleasebranch
> :
>
>
>
> My question is how do I activate other modules?  In previous versions I
> would have amended the moudules conf file and re-compile.  I can't find the
> modules file in this install
>
>
>
>
>
> thanks
>
> Matt
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From daveh at beachdognet.com  Thu Jun  9 00:44:48 2016
From: daveh at beachdognet.com (Dave Horton)
Date: Wed, 8 Jun 2016 16:44:48 -0400
Subject: [Freeswitch-users] Does mod_rss even work?
Message-ID: <E05B194D-8968-4291-AC1A-1DE360DAD5ED@beachdognet.com>

I?m trying to get a basic example mod_rss app to work, as per the example in https://wiki.freeswitch.org/wiki/Mod_rss, 
but running into multiple failures.

The current blocking problem is that mod_rss is complaining of ill-formed xml syntax in the downloaded rss document:

2016-06-08 16:21:08.815054 [WARNING] mod_rss.c:414 Error [[error near line 553]: unexpected closing tag </rss>]

However, I have inspected the document (the specific document is available at http://www.dansdeals.com/feed) and it is valid syntax, 
and there is in fact an opening <rss> tag in the document.

Before this, I ran into several problems that I had to hack the code to even get the basic menu to play and detect dtmf properly:

1) The Main Menu text was putting in angle brackets into the text for some reason, causing them to get read out as ?greater than?.  
Almost looks like someone thought they were formatting xml ?

         //switch_snprintf(buf + len, sizeof(buf) - len, "%d: %s. />", idx + 1, feed_names[idx]);

2) The buffer into which dtmf was being collected was not properly zeroed out, so that dtmf entry of ?1? got treated as ?1000? by mod_rss, causing the session to end.

I am running FreeSWITCH Version 1.6.8+git~20160505T153832Z~99de0ad502~64bit (git 99de0ad 2016-05-05 15:38:32Z 64bit)

Either I am doing something wrong, or this doesn?t seem to work at all?..

Dave

From alhakeem at gmail.com  Thu Jun  9 01:02:37 2016
From: alhakeem at gmail.com (Abdul Hakeem)
Date: Wed, 8 Jun 2016 22:02:37 +0100
Subject: [Freeswitch-users] XML bindings config examples via mod_event_0MQ
Message-ID: <CAO=waFBrKjBtwNmQgypVW_b12ksMtngY2Vs1LqBPsCZPiZxnxQ@mail.gmail.com>

Hello,

Could anyone point me to xml bindigs config examples via ZeroMQ (similar to
mod_xml_curl)?.

Cheers,
AH
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From matthew at brightfire.net  Thu Jun  9 01:05:02 2016
From: matthew at brightfire.net (Matthew)
Date: Wed, 8 Jun 2016 16:05:02 -0500
Subject: [Freeswitch-users] Problems with wait_for_silence API call
Message-ID: <fbd83d1b-6309-e170-25f4-a62f73428802@brightfire.net>

Hello All,

I've having a problem with the wait_for_silence API call in lua. When I 
attempt to orginate a call and then check for silence on the line, I'm 
getting unpredictable results. Sometimes it seems to work, but more 
times than not it times out ( switch_ivr_wait_for_silence: TIMEOUT 0 ). 
To test this, I answer the call, speak a short greeting and then mute 
the handset. In other words, I know the script should be detecting 
silence on the other end of the call. Here is a good portion of the test 
script I'm using ...

local cid_name   = argv[1]
local cid_number = argv[2]
local gw_string  = argv[3]
local queue_ext  = argv[4]

dialout_string =
   "{origination_caller_id_name='" .. cid_name .. "'," ..
   "origination_caller_id_number=" .. cid_number .. "," ..
   "ignore_early_media=true}" .. gw_string

dialout_session = freeswitch.Session( dialout_string )

while dialout_session:ready() and not dialout_session:answered() do
   freeswitch.msleep( 250 )
end

if dialout_session:ready() and dialout_session:answered() then

   dialout_session:consoleLog( "INFO", "dialout: connected to " .. 
gw_string ) )

   dialout_session:execute( "voice_start" )
   dialout_session:sleep( 2000 )
   dialout_session:execute( "wait_for_silence", "300 30 15 15000" )
   dialout_session:execute( "voice_stop" )

   dialout_session:consoleLog( "INFO", "dialout: amd_detect = " .. 
amd_detect )

end

And here is an example of how I would invoke the script ...

luarun dialout.lua TEST 1234567890 sofia/gateway/twilio/+1234567890 7000

Any ideas why the wait_for_silence API call would be misbehaving? Any 
suggestions on how I could debug this problem further?

Thanks,

-Matthew


From mike at jerris.com  Thu Jun  9 01:10:41 2016
From: mike at jerris.com (Michael Jerris)
Date: Wed, 8 Jun 2016 17:10:41 -0400
Subject: [Freeswitch-users] XML bindings config examples via
	mod_event_0MQ
In-Reply-To: <CAO=waFBrKjBtwNmQgypVW_b12ksMtngY2Vs1LqBPsCZPiZxnxQ@mail.gmail.com>
References: <CAO=waFBrKjBtwNmQgypVW_b12ksMtngY2Vs1LqBPsCZPiZxnxQ@mail.gmail.com>
Message-ID: <2B8440C3-428E-4C99-B545-D2198D1CFF36@jerris.com>

I STRONGLY do NOT recommend using zeromq for anything that is in freeswitch process.  The library will cause freeswitch to segfault.


> On Jun 8, 2016, at 5:02 PM, Abdul Hakeem <alhakeem at gmail.com> wrote:
> 
> Hello,
> 
> Could anyone point me to xml bindigs config examples via ZeroMQ (similar to mod_xml_curl)?.
> 
> Cheers,
> AH
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From anthony.minessale at gmail.com  Thu Jun  9 02:53:58 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 8 Jun 2016 17:53:58 -0500
Subject: [Freeswitch-users] Does mod_rss even work?
In-Reply-To: <E05B194D-8968-4291-AC1A-1DE360DAD5ED@beachdognet.com>
References: <E05B194D-8968-4291-AC1A-1DE360DAD5ED@beachdognet.com>
Message-ID: <CAKbxfG_+mTVZhQOP39c8CMBHYNfPq0ZPvHi4rxSmT33QPwu+Cw@mail.gmail.com>

It seems like you answered your own question if its not working.
This module has fallen into disuse and you should file a JIRA
https://freeswitch.org/jira
Provide examples and steps to reproduce.


On Wed, Jun 8, 2016 at 3:44 PM, Dave Horton <daveh at beachdognet.com> wrote:

> I?m trying to get a basic example mod_rss app to work, as per the example
> in https://wiki.freeswitch.org/wiki/Mod_rss,
> but running into multiple failures.
>
> The current blocking problem is that mod_rss is complaining of ill-formed
> xml syntax in the downloaded rss document:
>
> 2016-06-08 16:21:08.815054 [WARNING] mod_rss.c:414 Error [[error near line
> 553]: unexpected closing tag </rss>]
>
> However, I have inspected the document (the specific document is available
> at http://www.dansdeals.com/feed) and it is valid syntax,
> and there is in fact an opening <rss> tag in the document.
>
> Before this, I ran into several problems that I had to hack the code to
> even get the basic menu to play and detect dtmf properly:
>
> 1) The Main Menu text was putting in angle brackets into the text for some
> reason, causing them to get read out as ?greater than?.
> Almost looks like someone thought they were formatting xml ?
>
>          //switch_snprintf(buf + len, sizeof(buf) - len, "%d: %s. />", idx
> + 1, feed_names[idx]);
>
> 2) The buffer into which dtmf was being collected was not properly zeroed
> out, so that dtmf entry of ?1? got treated as ?1000? by mod_rss, causing
> the session to end.
>
> I am running FreeSWITCH Version
> 1.6.8+git~20160505T153832Z~99de0ad502~64bit (git 99de0ad 2016-05-05
> 15:38:32Z 64bit)
>
> Either I am doing something wrong, or this doesn?t seem to work at all?..
>
> Dave
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From cody.rosenbloom at gmail.com  Thu Jun  9 02:54:17 2016
From: cody.rosenbloom at gmail.com (Cody Rosenbloom)
Date: Wed, 8 Jun 2016 18:54:17 -0400
Subject: [Freeswitch-users] Testing for call hangup in dialplan xml
Message-ID: <CAPrxTnqAneapO9Zpfx73sMGOGdV9mFrtZaEjgmV_rjzhG-gn0g@mail.gmail.com>

Hi,

Is there a way to check if call has been hung up in dialplan xml. I want to
run lua script to process voicemail after caller has hungup the call. I
tried checking if state is equal to CS_HANGUP but it does not work. Current
dialplan xml is at http://pastebin.com/5PMKSrdh

Thanks
Cody
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From krice at freeswitch.org  Thu Jun  9 02:59:30 2016
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 8 Jun 2016 17:59:30 -0500
Subject: [Freeswitch-users] add modules when installing from the latest
	branch
In-Reply-To: <CAEeV61h+5Gy2Wo16jOEze2qPGmrD8q8pb3q01wTViAv2Sz+Z7A@mail.gmail.com>
References: <CAEeV61jv_rX-ebF6O4gM3=DG3jEDdMaBBovYtj_2QAwzozPJrg@mail.gmail.com>
	<CAF7jizJ7yXS5a0QhFm5uY1vZttcmO5HoM3sssCAqLY1NJyO37g@mail.gmail.com>
	<CAEeV61g12Ar2euivZQb+3NkTB7mRO7keRe12AN7kp9J8Do3xZw@mail.gmail.com>
	<007501d1c0d2$2febc220$8fc34660$@freeswitch.org>
	<CAEeV61h+5Gy2Wo16jOEze2qPGmrD8q8pb3q01wTViAv2Sz+Z7A@mail.gmail.com>
Message-ID: <B670C6AB-D0AD-4A17-AAF2-F8377F88C43D@freeswitch.org>

With this method of install everything is in an fhs compliant location. You can mod the commandline args in he systemd files tho

Sent from my iPhone

> On Jun 8, 2016, at 3:21 PM, Matt Broad <matt at supportedbusiness.com> wrote:
> 
> thanks Ken, just another quick question on this install method; how would I go about changing the default locations for the scripts? 
> 
> I would ordinarily use the following: ./configure --with-scriptdir=/etc/freeswitch/scripts/
> 
> 
> 
> thanks
> Matt
> 
>> On 7 June 2016 at 16:35, Ken Rice <krice at freeswitch.org> wrote:
>> Easier is probably just apt-cache search freeswitch or apt-cache search freeswitch|grep mongo or something like that
>> 
>>  
>> 
>> Once you do that you can apt-get install the module you want and then you just need to enable it in your /etc/freeswitch/autoload_configs/modules.conf.xml
>> 
>>  
>> 
>> The entire package system is designed so that you can use https://freeswitch.org/stash/projects/FS/repos/freeswitch-config-skel/browse to generate your own configuration package with deps on the modules that you want to include and just install that and bam you?re good
>> 
>>  
>> 
>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Matt Broad
>> Sent: Tuesday, June 7, 2016 5:50 AM
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Subject: Re: [Freeswitch-users] add modules when installing from the latest branch
>> 
>>  
>> 
>> perfect! 
>> 
>>  
>> 
>> thanks for the quick response.  Is this listed on the confluence site anywhere?  I am happy to add this if given access
>> 
>>  
>> 
>> thanks
>> 
>> Matt
>> 
>>  
>> 
>> On 7 June 2016 at 11:35, Benjamin Cropley <benjamin.cropley at gmail.com> wrote:
>> 
>> Say you wanted to install mod_mongo..
>> 
>>  
>> 
>> you can find the packages here http://files.freeswitch.org/repo/deb/freeswitch-1.6/dists/jessie/main/binary-amd64/Packages
>> 
>>  
>> 
>> apt-get install -y freeswitch-mod-mongo
>> 
>>  
>> 
>> then edit the modules conf as you would have done previously
>> 
>>  
>> 
>> On Tue, Jun 7, 2016 at 11:20 AM, Matt Broad <matt at supportedbusiness.com> wrote:
>> 
>> Hi,
>> 
>>  
>> 
>> this may be a really stupid question but here goes:
>> 
>>  
>> 
>> I have upgraded a server to debian 8 and have installed freeswitch using the latest branch
>> 
>> https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie#Debian8Jessie-latestreleasebranch:
>> 
>>  
>> 
>> My question is how do I activate other modules?  In previous versions I would have amended the moudules conf file and re-compile.  I can't find the modules file in this install 
>> 
>>  
>> 
>>  
>> 
>> thanks
>> 
>> Matt
>> 
>>  
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>>  
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>>  
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From krice at freeswitch.org  Thu Jun  9 03:01:53 2016
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 8 Jun 2016 18:01:53 -0500
Subject: [Freeswitch-users] Testing for call hangup in dialplan xml
In-Reply-To: <CAPrxTnqAneapO9Zpfx73sMGOGdV9mFrtZaEjgmV_rjzhG-gn0g@mail.gmail.com>
References: <CAPrxTnqAneapO9Zpfx73sMGOGdV9mFrtZaEjgmV_rjzhG-gn0g@mail.gmail.com>
Message-ID: <6839235B-178F-43C9-8D17-5D740D2DB17D@freeswitch.org>

This is why we have things like CDRs that can trigger post call processing via web call or via file drop. If you are using mod_voicemail there is a post voicemail being left hook already exposed in the configs.

Doing post call processing on the channel greatly increases the chances of hung channels resulting in system resources being held for dead calls

Sent from my iPhone

> On Jun 8, 2016, at 5:54 PM, Cody Rosenbloom <cody.rosenbloom at gmail.com> wrote:
> 
> Hi,
> 
> Is there a way to check if call has been hung up in dialplan xml. I want to run lua script to process voicemail after caller has hungup the call. I tried checking if state is equal to CS_HANGUP but it does not work. Current dialplan xml is at http://pastebin.com/5PMKSrdh
> 
> Thanks
> Cody
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From anthony.minessale at gmail.com  Thu Jun  9 03:33:45 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 8 Jun 2016 18:33:45 -0500
Subject: [Freeswitch-users] Check out ClueCon weekly featuring Amandine
 LePape and Matthew Hodgson from matrix.org
Message-ID: <CAKbxfG93P1+LbEhQpJoAHM7kJ9xaD+mYB1H5LPUtTkLQ7Jw0eQ@mail.gmail.com>

Interesting use of FreeSWITCH

https://www.youtube.com/watch?v=nF93l0DbAg0


-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From andrew.keil at visytel.com  Thu Jun  9 11:17:25 2016
From: andrew.keil at visytel.com (Andrew Keil)
Date: Thu, 9 Jun 2016 07:17:25 +0000
Subject: [Freeswitch-users] Re- Sending "api <conferencename> hup all" after
 Lua service script detects local hangup
Message-ID: <ME1PR01MB091540844AE7ADCA600D8B14F35F0@ME1PR01MB0915.ausprd01.prod.outlook.com>

To FreeSWITCH Users,

I have just come across an issue when my Lua script hangs up while inside a conference (let's call it conference name: "testconf").  I would like to be able to terminate everyone in the conference when this happens, however when I try to call session:execute("conference", "testconf hup all") nothing happens.  There is some mention of zombie_exec setting inside the DEBUG logging, however I have no idea where that is and how that is used.

If someone has a neat trick to enable this to work I would appreciate it.

Kind Regards,

Andrew Keil
Visytel Pty Ltd

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From gmaruzz at gmail.com  Thu Jun  9 11:26:11 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 9 Jun 2016 09:26:11 +0200
Subject: [Freeswitch-users] Re- Sending "api <conferencename> hup all"
 after Lua service script detects local hangup
In-Reply-To: <ME1PR01MB091540844AE7ADCA600D8B14F35F0@ME1PR01MB0915.ausprd01.prod.outlook.com>
References: <ME1PR01MB091540844AE7ADCA600D8B14F35F0@ME1PR01MB0915.ausprd01.prod.outlook.com>
Message-ID: <CALXCt0rCCUs7CViGR=et67P81DVQTWHdhn3vn3oC6XFLEWPZEQ@mail.gmail.com>

At that point you probably have an invalid "session" object.

I would try an hangup hook, that is executed after hangup in a freshly
spawned process.


another way is, earlier in your script, create an API object, then use that
API object to terminate the conference.

-giovanni

On Thu, Jun 9, 2016 at 9:17 AM, Andrew Keil <andrew.keil at visytel.com> wrote:

> To FreeSWITCH Users,
>
>
>
> I have just come across an issue when my Lua script hangs up while inside
> a conference (let?s call it conference name: ?testconf?).  I would like to
> be able to terminate everyone in the conference when this happens, however
> when I try to call session:execute(?conference?, ?testconf hup all?)
> nothing happens.  There is some mention of zombie_exec setting inside the
> DEBUG logging, however I have no idea where that is and how that is used.
>
>
>
> If someone has a neat trick to enable this to work I would appreciate it.
>
>
>
> Kind Regards,
>
>
>
> Andrew Keil
>
> *Visytel Pty Ltd*
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From andrew.keil at visytel.com  Thu Jun  9 11:40:44 2016
From: andrew.keil at visytel.com (Andrew Keil)
Date: Thu, 9 Jun 2016 07:40:44 +0000
Subject: [Freeswitch-users] Re- Sending "api <conferencename> hup all"
 after Lua service script detects local hangup
In-Reply-To: <CALXCt0rCCUs7CViGR=et67P81DVQTWHdhn3vn3oC6XFLEWPZEQ@mail.gmail.com>
References: <ME1PR01MB091540844AE7ADCA600D8B14F35F0@ME1PR01MB0915.ausprd01.prod.outlook.com>
	<CALXCt0rCCUs7CViGR=et67P81DVQTWHdhn3vn3oC6XFLEWPZEQ@mail.gmail.com>
Message-ID: <ME1PR01MB0915CFAB6CC6B0F17193DC10F35F0@ME1PR01MB0915.ausprd01.prod.outlook.com>

Thanks for you fast response.  When you say API object do you mean something like this:

Early in my Lua script:
api = freeswitch.API()

After HUNGUP detected:
api:execute(?conference?, ?testconf hup all?)

Do I then have to dispose of the api object in some way prior to the Lua script exiting?  Or is this done automatically?

Thanks,

Andrew

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli
Sent: Thursday, 9 June 2016 5:26 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Re- Sending "api <conferencename> hup all" after Lua service script detects local hangup

At that point you probably have an invalid "session" object.
I would try an hangup hook, that is executed after hangup in a freshly spawned process.

another way is, earlier in your script, create an API object, then use that API object to terminate the conference.
-giovanni

On Thu, Jun 9, 2016 at 9:17 AM, Andrew Keil <andrew.keil at visytel.com<mailto:andrew.keil at visytel.com>> wrote:
To FreeSWITCH Users,

I have just come across an issue when my Lua script hangs up while inside a conference (let?s call it conference name: ?testconf?).  I would like to be able to terminate everyone in the conference when this happens, however when I try to call session:execute(?conference?, ?testconf hup all?) nothing happens.  There is some mention of zombie_exec setting inside the DEBUG logging, however I have no idea where that is and how that is used.

If someone has a neat trick to enable this to work I would appreciate it.

Kind Regards,

Andrew Keil
Visytel Pty Ltd


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

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

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



--
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From gmaruzz at gmail.com  Thu Jun  9 11:54:19 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 9 Jun 2016 09:54:19 +0200
Subject: [Freeswitch-users] Re- Sending "api <conferencename> hup all"
 after Lua service script detects local hangup
In-Reply-To: <ME1PR01MB0915CFAB6CC6B0F17193DC10F35F0@ME1PR01MB0915.ausprd01.prod.outlook.com>
References: <ME1PR01MB091540844AE7ADCA600D8B14F35F0@ME1PR01MB0915.ausprd01.prod.outlook.com>
	<CALXCt0rCCUs7CViGR=et67P81DVQTWHdhn3vn3oC6XFLEWPZEQ@mail.gmail.com>
	<ME1PR01MB0915CFAB6CC6B0F17193DC10F35F0@ME1PR01MB0915.ausprd01.prod.outlook.com>
Message-ID: <CALXCt0qWRRsVGS=3vqRrq8stgkfe3JOssWzN70FxuFQqUgLEqQ@mail.gmail.com>

On Thu, Jun 9, 2016 at 9:40 AM, Andrew Keil <andrew.keil at visytel.com> wrote:

> Thanks for you fast response.  When you say API object do you mean
> something like this:
>
>
>
> Early in my Lua script:
>
> api = freeswitch.API()
>
>
>
> After HUNGUP detected:
>
> api:execute(?conference?, ?testconf hup all?)
>

yes


>
>
> Do I then have to dispose of the api object in some way prior to the Lua
> script exiting?  Or is this done automatically?
>

is done automatically


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From alex at digitalmail.com  Thu Jun  9 14:47:59 2016
From: alex at digitalmail.com (Alex Lake)
Date: Thu, 09 Jun 2016 10:47:59 +0000
Subject: [Freeswitch-users] 1.4 dialplans on 1.6
Message-ID: <ac7b4e1e-0d74-0fa1-361d-52d6846784f3@digitalmail.com>

Hello....

It's been a while since I've installed Freeswitch. We had just taken the 
plunge with 1.4 on Ubuntu 12.04!

Now I'm wondering if 1.6 on Debian 8 would be compatible with all our 
dialplans and Lua scripts.

Any ideas?


From vittico at gmail.com  Thu Jun  9 15:35:26 2016
From: vittico at gmail.com (Victor Medina)
Date: Thu, 9 Jun 2016 07:35:26 -0400
Subject: [Freeswitch-users] 1.4 dialplans on 1.6
In-Reply-To: <ac7b4e1e-0d74-0fa1-361d-52d6846784f3@digitalmail.com>
References: <ac7b4e1e-0d74-0fa1-361d-52d6846784f3@digitalmail.com>
Message-ID: <CAEA-bfF7iho5hKFhJPKw10h8C-eug7YtqXrtONcO+ZC9yNJpXg@mail.gmail.com>

Hi Alex.

There should be no problems.  As usual some testing might be a good Idea.


Sin mas a que hacer referencia,

Victor Medina

On Thu, Jun 9, 2016 at 6:47 AM, Alex Lake <alex at digitalmail.com> wrote:

> Hello....
>
> It's been a while since I've installed Freeswitch. We had just taken the
> plunge with 1.4 on Ubuntu 12.04!
>
> Now I'm wondering if 1.6 on Debian 8 would be compatible with all our
> dialplans and Lua scripts.
>
> Any ideas?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From yu at yu-boot.ru  Thu Jun  9 15:29:53 2016
From: yu at yu-boot.ru (Yu Boot)
Date: Thu, 9 Jun 2016 14:29:53 +0300
Subject: [Freeswitch-users] SIP phone displays b_callee_name and
	b_callee_num on outgoing calls
Message-ID: <d5ce5439-2762-55ae-8b5a-4479857ccda9@yu-boot.ru>

Hello.

Help me please.

When I try to dial from Cisco SPA504G SIP phone via FS, after successful 
connection I see on screen "Outbound Call" and number at ip, which are 
values of variables b_callee_name and b_callee_num. I've never seen 
something like that on any other PBX (Asterisk, Yate, proprietary 
solutions). How to disable this behavior? It's not only confusing, it 
prevents me from simple redial lasy dialed number by respective button, 
as SIP phone tries to dial to that b_callee_num, which is in some 
different format.



From fanx07 at gmail.com  Thu Jun  9 13:31:07 2016
From: fanx07 at gmail.com (Anonim Stefan)
Date: Thu, 9 Jun 2016 12:31:07 +0300
Subject: [Freeswitch-users] Group calls without REGISTER users
Message-ID: <CAO7wsJt=tCLgi-f0_gAKqjWLsRMR+UZhhYVJzm+8e_8cNG-7eQ@mail.gmail.com>

Hi,

I'm a Freeswitch newbee and I'm trying to make a group conference call
setup:

Kamailio sends Group-Users:user1000;ip1000;.. ----------> Freeswitch

I've seen that "conference_set_auto_outcall" does what I need, but the
users must be registered: "2016-06-09 08:46:52.161319 [ERR]
mod_conference.c:1389 Cannot create outgoing channel, cause:
USER_NOT_REGISTERED".

Without success, I tried setting:
-> <user id="1000" cidr="IP1000"> in directory (maybe this should solve it
and I'm doing something wrong)
-> <param name="disable-register" value="true"/> in sip_profiles

I can access the received header values in dialplan and get the user1000
and ip1000.

Is there a way I could tell Freeswitch to add user1000 at IP1000 to group
conference without needing to register them, just using the given IP?

Also, what should I use if I want to keep a local "IP1000<=>x.x.x.x"
mapping and avoid sending the actual IP in the Group-Users header?

Thank you in advance,
Stefan
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From fanx07 at gmail.com  Thu Jun  9 16:27:36 2016
From: fanx07 at gmail.com (Anonim Stefan)
Date: Thu, 9 Jun 2016 15:27:36 +0300
Subject: [Freeswitch-users] Group calls without REGISTER users
Message-ID: <CAO7wsJvFV1yw5riHq6k-Ysw5LL5S2mULsvR_7q6sTR1WrHfvwQ@mail.gmail.com>

Hi,

I'm a Freeswitch newbee and I'm trying to make a group conference call
setup:

Kamailio sends Group-Users:user1000;ip1000;.. ----------> Freeswitch

I've seen that "conference_set_auto_outcall" does what I need, but the
users must be registered: "2016-06-09 08:46:52.161319 [ERR]
mod_conference.c:1389 Cannot create outgoing channel, cause:
USER_NOT_REGISTERED".

Without success, I tried setting:
-> <user id="1000" cidr="IP1000"> in directory (maybe this should solve it
and I'm doing something wrong)
-> <param name="disable-register" value="true"/> in sip_profiles

I can access the received header values in dialplan and get the user1000
and ip1000.

Is there a way I could tell Freeswitch to add user1000 at IP1000 to group
conference without needing to register them, just using the given IP?

Also, what should I use if I want to keep a local "IP1000<=>x.x.x.x"
mapping and avoid sending the actual IP in the Group-Users header?

Thank you in advance,
Stefan
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From krice at freeswitch.org  Thu Jun  9 17:01:58 2016
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 9 Jun 2016 08:01:58 -0500
Subject: [Freeswitch-users] SIP phone displays b_callee_name and
	b_callee_num on outgoing calls
In-Reply-To: <d5ce5439-2762-55ae-8b5a-4479857ccda9@yu-boot.ru>
References: <d5ce5439-2762-55ae-8b5a-4479857ccda9@yu-boot.ru>
Message-ID: <D4663FAD-4E87-4EE2-A019-78BD0E51FD03@freeswitch.org>

Its called display update and its very useful for ipdating the callers display with caller id info of the called party. You can disable it per call or across the board with ignore display update set one the channel or set as a global var

Sent from my iPhone

> On Jun 9, 2016, at 6:29 AM, Yu Boot <yu at yu-boot.ru> wrote:
> 
> Hello.
> 
> Help me please.
> 
> When I try to dial from Cisco SPA504G SIP phone via FS, after successful 
> connection I see on screen "Outbound Call" and number at ip, which are 
> values of variables b_callee_name and b_callee_num. I've never seen 
> something like that on any other PBX (Asterisk, Yate, proprietary 
> solutions). How to disable this behavior? It's not only confusing, it 
> prevents me from simple redial lasy dialed number by respective button, 
> as SIP phone tries to dial to that b_callee_num, which is in some 
> different format.
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From lists at kavun.ch  Thu Jun  9 18:05:59 2016
From: lists at kavun.ch (Emrah)
Date: Thu, 9 Jun 2016 16:05:59 +0200
Subject: [Freeswitch-users] SIP phone displays b_callee_name and
	b_callee_num on outgoing calls
In-Reply-To: <d5ce5439-2762-55ae-8b5a-4479857ccda9@yu-boot.ru>
References: <d5ce5439-2762-55ae-8b5a-4479857ccda9@yu-boot.ru>
Message-ID: <C7F85245-7ADF-495E-A820-F16E1651306D@kavun.ch>

This is probably what you?re looking for:
ignore_display_updates 
Tells freeswitch not to send display UPDATEs to the leg of the call. (update_display)
Usage: From dialplan/default.xml:
To set on A-Leg
<action application="set"  data="ignore_display_updates=true"/>

To set on B-Leg
<action application="bridge" data="{ignore_display_updates=true}sofia/gateway/provider/18005551212"/>


> On Jun 9, 2016, at 1:29 PM, Yu Boot <yu at yu-boot.ru> wrote:
> 
> Hello.
> 
> Help me please.
> 
> When I try to dial from Cisco SPA504G SIP phone via FS, after successful 
> connection I see on screen "Outbound Call" and number at ip, which are 
> values of variables b_callee_name and b_callee_num. I've never seen 
> something like that on any other PBX (Asterisk, Yate, proprietary 
> solutions). How to disable this behavior? It's not only confusing, it 
> prevents me from simple redial lasy dialed number by respective button, 
> as SIP phone tries to dial to that b_callee_num, which is in some 
> different format.
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From alex at digitalmail.com  Thu Jun  9 18:13:53 2016
From: alex at digitalmail.com (Alex Lake)
Date: Thu, 09 Jun 2016 14:13:53 +0000
Subject: [Freeswitch-users] 1.4 dialplans on 1.6
In-Reply-To: <CAEA-bfF7iho5hKFhJPKw10h8C-eug7YtqXrtONcO+ZC9yNJpXg@mail.gmail.com>
References: <ac7b4e1e-0d74-0fa1-361d-52d6846784f3@digitalmail.com>
	<CAEA-bfF7iho5hKFhJPKw10h8C-eug7YtqXrtONcO+ZC9yNJpXg@mail.gmail.com>
Message-ID: <d20a1e57-1943-3a65-fc15-b5bfec0a8ad9@digitalmail.com>

That's great news. Just getting the server ordered now....

On 09/06/2016 12:35, Victor Medina wrote:
> Hi Alex.
>
> There should be no problems.  As usual some testing might be a good Idea.
>
>
> Sin mas a que hacer referencia,
>
> Victor Medina
>
> On Thu, Jun 9, 2016 at 6:47 AM, Alex Lake <alex at digitalmail.com 
> <mailto:alex at digitalmail.com>> wrote:
>
>     Hello....
>
>     It's been a while since I've installed Freeswitch. We had just
>     taken the
>     plunge with 1.4 on Ubuntu 12.04!
>
>     Now I'm wondering if 1.6 on Debian 8 would be compatible with all our
>     dialplans and Lua scripts.
>
>     Any ideas?
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



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From silviu.cpp at gmail.com  Thu Jun  9 19:28:19 2016
From: silviu.cpp at gmail.com (Caragea Silviu)
Date: Thu, 9 Jun 2016 18:28:19 +0300
Subject: [Freeswitch-users] Recording a bridge call
Message-ID: <CAJsc=ovcxNSiCY-tg+_bKYoYPhs5TPHHtDNP36Cp0rPVd_rbQw@mail.gmail.com>

Hello,

I want to start recording a call once it's bridged (the other party
answered). And stop recording once the other party closed the call.

I'm currently using uuid_record but It's recording everything including the
ringtone and also after the other party closed the call all other messages
I'm playing to the caller.

Silviu
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From guy.callum at gmail.com  Thu Jun  9 19:31:49 2016
From: guy.callum at gmail.com (Callum Guy)
Date: Thu, 09 Jun 2016 15:31:49 +0000
Subject: [Freeswitch-users] Recording a bridge call
In-Reply-To: <CAJsc=ovcxNSiCY-tg+_bKYoYPhs5TPHHtDNP36Cp0rPVd_rbQw@mail.gmail.com>
References: <CAJsc=ovcxNSiCY-tg+_bKYoYPhs5TPHHtDNP36Cp0rPVd_rbQw@mail.gmail.com>
Message-ID: <CAM3_A=oSHNn_z3uZw8BOOjmQxqYEci_b2U1BnukeawFBoJK-rQ@mail.gmail.com>

You'll want a little something like this (on your B-leg):

<action application=?set?
data=?record_file_name={recordings_dir}/${strftime(%Y-%m-%d-%H-%M-%S)}_${uuid}.wav?
inline=?true?/><action application=?export?
data=?execute_on_answer=record_session {record_file_name}?/>

Trusty "execute on answer"!

On Thu, 9 Jun 2016 at 16:29 Caragea Silviu <silviu.cpp at gmail.com> wrote:

> Hello,
>
> I want to start recording a call once it's bridged (the other party
> answered). And stop recording once the other party closed the call.
>
> I'm currently using uuid_record but It's recording everything including
> the ringtone and also after the other party closed the call all other
> messages I'm playing to the caller.
>
> Silviu
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From giggsey at gmail.com  Thu Jun  9 19:35:32 2016
From: giggsey at gmail.com (Joshua Gigg)
Date: Thu, 09 Jun 2016 15:35:32 +0000
Subject: [Freeswitch-users] Recording a bridge call
In-Reply-To: <CAM3_A=oSHNn_z3uZw8BOOjmQxqYEci_b2U1BnukeawFBoJK-rQ@mail.gmail.com>
References: <CAJsc=ovcxNSiCY-tg+_bKYoYPhs5TPHHtDNP36Cp0rPVd_rbQw@mail.gmail.com>
	<CAM3_A=oSHNn_z3uZw8BOOjmQxqYEci_b2U1BnukeawFBoJK-rQ@mail.gmail.com>
Message-ID: <CADcg8iVBMTnHL+ENL3a5Scz+SzSYNpTq1nSzLZFPC8udjah90Q@mail.gmail.com>

If you are using uuid_record, you can use the 'media_bug_answer_req=true'
 variable to start recording once the channel has been answered.

On Thu, 9 Jun 2016 at 16:33 Callum Guy <guy.callum at gmail.com> wrote:

> You'll want a little something like this (on your B-leg):
>
> <action application=?set?
> data=?record_file_name={recordings_dir}/${strftime(%Y-%m-%d-%H-%M-%S)}_${uuid}.wav?
> inline=?true?/><action application=?export?
> data=?execute_on_answer=record_session {record_file_name}?/>
>
> Trusty "execute on answer"!
>
> On Thu, 9 Jun 2016 at 16:29 Caragea Silviu <silviu.cpp at gmail.com> wrote:
>
>> Hello,
>>
>> I want to start recording a call once it's bridged (the other party
>> answered). And stop recording once the other party closed the call.
>>
>> I'm currently using uuid_record but It's recording everything including
>> the ringtone and also after the other party closed the call all other
>> messages I'm playing to the caller.
>>
>> Silviu
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From pashdown at xmission.com  Thu Jun  9 19:54:29 2016
From: pashdown at xmission.com (Pete Ashdown)
Date: Thu, 9 Jun 2016 09:54:29 -0600
Subject: [Freeswitch-users] System global mode variable
In-Reply-To: <5756DF3E.8000004@xmission.com>
References: <5756DF3E.8000004@xmission.com>
Message-ID: <57599135.50908@xmission.com>

Answered my own question:

In your dialplan:
<action application="lua" data="script.lua"/>

In script.lua:
mode = freeswitch.getGlobalVariable("mode")
if (mode == "open") then
  session:speak("open")
end
if (mode == "techonly") then
  session:speak("teck only")
end
if (mode == "event") then
  session:speak("event")
else
  session:speak("i dont know")
end

Make sure the global is created somewhere, like vars.xml:
<X-PRE-PROCESS cmd="set" data="mode=open"/>


On 06/07/2016 08:50 AM, Pete Ashdown wrote:
> What is the simplest way to have a conditional branch based on a global
> variable in a dialplan?  Actually using a global variable and xml is
> giving me mixed results.  Do I need to use Lua or another scripting
> language?
>
> This is what I have, which doesn't give the results desired.   The
> "speak" actions are merely debug placeholders.  I know there are better
> ways to do conditional voice to text.
>
>       <condition regex="any" field="${cond(${mode} == 'open' ? YES :
> NO)}" expression="^YES$">
>             <action application="speak" data="open"/>
>       </condition>
>
>       <condition regex="any" field="${cond(${mode} == 'tech only' ? YES
> : NO)}" expression="^YES$">
>             <action application="speak" data="teck only"/>
>       </condition>
>
>       <condition regex="any" field="${cond(${mode} == 'event' ? YES :
> NO)}" expression="^YES$">
>             <action application="speak" data="event"/>
>       </condition>
>


From yu at yu-boot.ru  Thu Jun  9 17:07:43 2016
From: yu at yu-boot.ru (Yu Boot)
Date: Thu, 9 Jun 2016 16:07:43 +0300
Subject: [Freeswitch-users] SIP phone displays b_callee_name and
 b_callee_num on outgoing calls
In-Reply-To: <D4663FAD-4E87-4EE2-A019-78BD0E51FD03@freeswitch.org>
References: <d5ce5439-2762-55ae-8b5a-4479857ccda9@yu-boot.ru>
	<D4663FAD-4E87-4EE2-A019-78BD0E51FD03@freeswitch.org>
Message-ID: <21d24677-a5fe-af5d-5f28-cc51aae82efb@yu-boot.ru>

Thanks, it was send-display-update variable


09.06.2016 16:01, Ken Rice ?????:
> Its called display update and its very useful for ipdating the callers display with caller id info of the called party. You can disable it per call or across the board with ignore display update set one the channel or set as a global var
>
> Sent from my iPhone
>
>> On Jun 9, 2016, at 6:29 AM, Yu Boot <yu at yu-boot.ru> wrote:
>>
>> Hello.
>>
>> Help me please.
>>
>> When I try to dial from Cisco SPA504G SIP phone via FS, after successful
>> connection I see on screen "Outbound Call" and number at ip, which are
>> values of variables b_callee_name and b_callee_num. I've never seen
>> something like that on any other PBX (Asterisk, Yate, proprietary
>> solutions). How to disable this behavior? It's not only confusing, it
>> prevents me from simple redial lasy dialed number by respective button,
>> as SIP phone tries to dial to that b_callee_num, which is in some
>> different format.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From silviu.cpp at gmail.com  Thu Jun  9 22:11:27 2016
From: silviu.cpp at gmail.com (Caragea Silviu)
Date: Thu, 9 Jun 2016 21:11:27 +0300
Subject: [Freeswitch-users] Recording a bridge call
In-Reply-To: <CADcg8iVBMTnHL+ENL3a5Scz+SzSYNpTq1nSzLZFPC8udjah90Q@mail.gmail.com>
References: <CAJsc=ovcxNSiCY-tg+_bKYoYPhs5TPHHtDNP36Cp0rPVd_rbQw@mail.gmail.com>
	<CAM3_A=oSHNn_z3uZw8BOOjmQxqYEci_b2U1BnukeawFBoJK-rQ@mail.gmail.com>
	<CADcg8iVBMTnHL+ENL3a5Scz+SzSYNpTq1nSzLZFPC8udjah90Q@mail.gmail.com>
Message-ID: <CAJsc=oubiSU6PXL6ZC=DmZ0O8bPHdxZ+LJJdrCPYk8vq6gwG=w@mail.gmail.com>

So basically what I have is :

Leg A is answered and I play some ivr message inside then I'm bridging this
leg by calling another number (and this is the call I want to record). Once
the other side is closing the call I want to stop the recording.

media_bug_answer_req I don't think will work as time the call is already
answered.

Silviu

On Thu, Jun 9, 2016 at 6:35 PM, Joshua Gigg <giggsey at gmail.com> wrote:

> If you are using uuid_record, you can use the 'media_bug_answer_req=true'
>  variable to start recording once the channel has been answered.
>
> On Thu, 9 Jun 2016 at 16:33 Callum Guy <guy.callum at gmail.com> wrote:
>
>> You'll want a little something like this (on your B-leg):
>>
>> <action application=?set?
>> data=?record_file_name={recordings_dir}/${strftime(%Y-%m-%d-%H-%M-%S)}_${uuid}.wav?
>> inline=?true?/><action application=?export?
>> data=?execute_on_answer=record_session {record_file_name}?/>
>>
>> Trusty "execute on answer"!
>>
>> On Thu, 9 Jun 2016 at 16:29 Caragea Silviu <silviu.cpp at gmail.com> wrote:
>>
>>> Hello,
>>>
>>> I want to start recording a call once it's bridged (the other party
>>> answered). And stop recording once the other party closed the call.
>>>
>>> I'm currently using uuid_record but It's recording everything including
>>> the ringtone and also after the other party closed the call all other
>>> messages I'm playing to the caller.
>>>
>>> Silviu
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From aqsyounas at gmail.com  Thu Jun  9 22:28:47 2016
From: aqsyounas at gmail.com (Aqs Younas)
Date: Thu, 9 Jun 2016 23:28:47 +0500
Subject: [Freeswitch-users] [ERR] helloworld.js:8 Exception: ReferenceError:
	File is not defined
Message-ID: <CAGVOjVXuXhdEbjrggx6aXmkADZ4Y1vXL0vEMaNLrwVfTdewdpg@mail.gmail.com>

Greetings list.

I am getting below error when i am trying open a file in mod_v8.

2016-06-02 01:40:44.908007 [ERR] helloworld.js:8 Exception: ReferenceError:
File is not defined (near: "        fd = new File(baseFolder + "/" + menuId
+ ".json");")

Any pointer is much appreciated

Thanks.
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From matthew at brightfire.net  Fri Jun 10 01:14:26 2016
From: matthew at brightfire.net (Matthew)
Date: Thu, 9 Jun 2016 16:14:26 -0500
Subject: [Freeswitch-users] Problems with wait_for_silence API call
In-Reply-To: <fbd83d1b-6309-e170-25f4-a62f73428802@brightfire.net>
References: <fbd83d1b-6309-e170-25f4-a62f73428802@brightfire.net>
Message-ID: <4eb0ca23-886c-72b8-da7e-a12cc78329e5@brightfire.net>

Hello All,

After poking around a bit more I realized is that wait_for_silence was 
never actually waiting for silence. It needs to detect non-silence 
before it will ever try to detect silence. That makes it quite useless 
if you call the function from a point where you can't guarantee that 
audio is being received :/ What I did was broke up the function into two 
smaller functions ...

detect_audio <threshold> <audio_hits> <timeout_ms> [<file>]
detect_silence <threshold> <silence_hits> <timeout_ms> [<file>]

This allowed me the flexibility I needed to build an outbound dial 
solution the way I wanted to. For example, I can now do ...

   amd_detect = "person"
   dialout_session:execute( "voice_start" )
   -- wait up to 3 seconds for 5 frames of audio above threshold 300
   dialout_session:execute( "detect_audio", "300 5 3000" )
   detect_audio_timeout = dialout_session:getVariable( 
"detect_audio_timeout" )
   dialout_session:consoleLog( "INFO", "dialout: detect_audio_timeout = 
" .. detect_audio_timeout )
   -- if audio was detected, wait up to 30 seconds for 30 frames of 
silence below threshold 300
   if detect_audio_timeout == "false" then
     dialout_session:execute( "detect_silence", "300 30 30000" )
   end
   dialout_session:execute( "voice_stop" )
   if( dialout_session:getVariable( "amd_status" ) ~= nil ) then
     amd_detect = dialout_session:getVariable( "amd_status" )
   end
   dialout_session:consoleLog( "INFO", "dialout: amd_detect = " .. 
amd_detect )
   if amd_detect == "machine" then
     ...
   else
    ...
   end

I have patches based on a current git repo. Should I submit those 
somewhere for inclusion upstream for other people to use or should I 
just maintain these locally?

Thanks,

-Matthew

On 6/8/2016 4:05 PM, Matthew wrote:
> Hello All,
>
> I've having a problem with the wait_for_silence API call in lua. When I
> attempt to orginate a call and then check for silence on the line, I'm
> getting unpredictable results. Sometimes it seems to work, but more
> times than not it times out ( switch_ivr_wait_for_silence: TIMEOUT 0 ).
> To test this, I answer the call, speak a short greeting and then mute
> the handset. In other words, I know the script should be detecting
> silence on the other end of the call. Here is a good portion of the test
> script I'm using ...
>
> local cid_name   = argv[1]
> local cid_number = argv[2]
> local gw_string  = argv[3]
> local queue_ext  = argv[4]
>
> dialout_string =
>     "{origination_caller_id_name='" .. cid_name .. "'," ..
>     "origination_caller_id_number=" .. cid_number .. "," ..
>     "ignore_early_media=true}" .. gw_string
>
> dialout_session = freeswitch.Session( dialout_string )
>
> while dialout_session:ready() and not dialout_session:answered() do
>     freeswitch.msleep( 250 )
> end
>
> if dialout_session:ready() and dialout_session:answered() then
>
>     dialout_session:consoleLog( "INFO", "dialout: connected to " ..
> gw_string ) )
>
>     dialout_session:execute( "voice_start" )
>     dialout_session:sleep( 2000 )
>     dialout_session:execute( "wait_for_silence", "300 30 15 15000" )
>     dialout_session:execute( "voice_stop" )
>
>     dialout_session:consoleLog( "INFO", "dialout: amd_detect = " ..
> amd_detect )
>
> end
>
> And here is an example of how I would invoke the script ...
>
> luarun dialout.lua TEST 1234567890 sofia/gateway/twilio/+1234567890 7000
>
> Any ideas why the wait_for_silence API call would be misbehaving? Any
> suggestions on how I could debug this problem further?
>
> Thanks,
>
> -Matthew
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mike at jerris.com  Fri Jun 10 02:00:59 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 9 Jun 2016 18:00:59 -0400
Subject: [Freeswitch-users] Problems with wait_for_silence API call
In-Reply-To: <4eb0ca23-886c-72b8-da7e-a12cc78329e5@brightfire.net>
References: <fbd83d1b-6309-e170-25f4-a62f73428802@brightfire.net>
	<4eb0ca23-886c-72b8-da7e-a12cc78329e5@brightfire.net>
Message-ID: <0B49608D-083E-4F32-96C9-176FC4E5A441@jerris.com>

Can you please submit a pull request for these changes so they can be reviewed?

> On Jun 9, 2016, at 5:14 PM, Matthew <matthew at brightfire.net> wrote:
> 
> Hello All,
> 
> After poking around a bit more I realized is that wait_for_silence was 
> never actually waiting for silence. It needs to detect non-silence 
> before it will ever try to detect silence. That makes it quite useless 
> if you call the function from a point where you can't guarantee that 
> audio is being received :/ What I did was broke up the function into two 
> smaller functions ...
> 
> detect_audio <threshold> <audio_hits> <timeout_ms> [<file>]
> detect_silence <threshold> <silence_hits> <timeout_ms> [<file>]
> 
> This allowed me the flexibility I needed to build an outbound dial 
> solution the way I wanted to. For example, I can now do ...
> 
>   amd_detect = "person"
>   dialout_session:execute( "voice_start" )
>   -- wait up to 3 seconds for 5 frames of audio above threshold 300
>   dialout_session:execute( "detect_audio", "300 5 3000" )
>   detect_audio_timeout = dialout_session:getVariable( 
> "detect_audio_timeout" )
>   dialout_session:consoleLog( "INFO", "dialout: detect_audio_timeout = 
> " .. detect_audio_timeout )
>   -- if audio was detected, wait up to 30 seconds for 30 frames of 
> silence below threshold 300
>   if detect_audio_timeout == "false" then
>     dialout_session:execute( "detect_silence", "300 30 30000" )
>   end
>   dialout_session:execute( "voice_stop" )
>   if( dialout_session:getVariable( "amd_status" ) ~= nil ) then
>     amd_detect = dialout_session:getVariable( "amd_status" )
>   end
>   dialout_session:consoleLog( "INFO", "dialout: amd_detect = " .. 
> amd_detect )
>   if amd_detect == "machine" then
>     ...
>   else
>    ...
>   end
> 
> I have patches based on a current git repo. Should I submit those 
> somewhere for inclusion upstream for other people to use or should I 
> just maintain these locally?
> 
> Thanks,
> 
> -Matthew
> 
> On 6/8/2016 4:05 PM, Matthew wrote:
>> Hello All,
>> 
>> I've having a problem with the wait_for_silence API call in lua. When I
>> attempt to orginate a call and then check for silence on the line, I'm
>> getting unpredictable results. Sometimes it seems to work, but more
>> times than not it times out ( switch_ivr_wait_for_silence: TIMEOUT 0 ).
>> To test this, I answer the call, speak a short greeting and then mute
>> the handset. In other words, I know the script should be detecting
>> silence on the other end of the call. Here is a good portion of the test
>> script I'm using ...
>> 
>> local cid_name   = argv[1]
>> local cid_number = argv[2]
>> local gw_string  = argv[3]
>> local queue_ext  = argv[4]
>> 
>> dialout_string =
>>    "{origination_caller_id_name='" .. cid_name .. "'," ..
>>    "origination_caller_id_number=" .. cid_number .. "," ..
>>    "ignore_early_media=true}" .. gw_string
>> 
>> dialout_session = freeswitch.Session( dialout_string )
>> 
>> while dialout_session:ready() and not dialout_session:answered() do
>>    freeswitch.msleep( 250 )
>> end
>> 
>> if dialout_session:ready() and dialout_session:answered() then
>> 
>>    dialout_session:consoleLog( "INFO", "dialout: connected to " ..
>> gw_string ) )
>> 
>>    dialout_session:execute( "voice_start" )
>>    dialout_session:sleep( 2000 )
>>    dialout_session:execute( "wait_for_silence", "300 30 15 15000" )
>>    dialout_session:execute( "voice_stop" )
>> 
>>    dialout_session:consoleLog( "INFO", "dialout: amd_detect = " ..
>> amd_detect )
>> 
>> end
>> 
>> And here is an example of how I would invoke the script ...
>> 
>> luarun dialout.lua TEST 1234567890 sofia/gateway/twilio/+1234567890 7000
>> 
>> Any ideas why the wait_for_silence API call would be misbehaving? Any
>> suggestions on how I could debug this problem further?
>> 
>> Thanks,
>> 
>> -Matthew



From daveh at beachdognet.com  Fri Jun 10 04:50:16 2016
From: daveh at beachdognet.com (Dave Horton)
Date: Thu, 9 Jun 2016 20:50:16 -0400
Subject: [Freeswitch-users] RFC 2833 payload type mismatch issue on bridged
	call
Message-ID: <B767942B-0074-4672-BCD3-4AD3DA5D9C99@beachdognet.com>

I have a client sending an INVITE to FS with RFC 2833 offered as payload type 96.  My freeswitch device profile has "rfc2833-pt" set to 101 and "inbound-late-negotiation? set to true.

The FS then bridges this call to a gateway, sending an INVITE offering pt 101 for 2833.
The 200 OK on the B leg comes back (with pt 101), and FS then sends a 200 OK to the original INVITE with pt 101.
So, to summarize, the client has offered 101 for 2833 and the FS responds with an SDP with 96.
At that point, DTMF does not work (DTMF generated by the client not recognized by the far end gateway).

Is there a configuration setting that would tell FS to use the RDC 2833 payload type from the A leg when bridging a call?

Interestingly, I am only seeing this problem on servers I have upgraded (to  1.6.8) ? older servers do bridge the B leg offering pt 96 if that came in on the A leg.
(Note, though, that the older servers are also not configured to do inbound-late-negotiation).

From daveh at beachdognet.com  Fri Jun 10 06:21:39 2016
From: daveh at beachdognet.com (Dave Horton)
Date: Thu, 9 Jun 2016 22:21:39 -0400
Subject: [Freeswitch-users] distributed presence (again)
Message-ID: <5B0BFA80-9BA4-46BD-AD70-5170F48D1CE8@beachdognet.com>

I?ve seen a few recent posts on distributed presence work, and I am running into the same need.
Like the other?s who have posted, I can?t use the same hostname across servers.
In my case, this is because in order to make call transfer scenarios work I need
to redirect calls sometimes to specific servers, and unique hostnames are required for that.

Is there another solution for this?  

If not, I?d be interested in working on a solution, as long as there was a view towards
incorporating it in the mainstream code at some point (i.e., I wouldn't want to have to 
keep patching releases going forward).

From krice at freeswitch.org  Fri Jun 10 07:09:57 2016
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 9 Jun 2016 22:09:57 -0500
Subject: [Freeswitch-users] distributed presence (again)
In-Reply-To: <5B0BFA80-9BA4-46BD-AD70-5170F48D1CE8@beachdognet.com>
References: <5B0BFA80-9BA4-46BD-AD70-5170F48D1CE8@beachdognet.com>
Message-ID: <D6F60388-9047-4221-8772-882052A80877@freeswitch.org>

Have you just using fs_path to send the call to a specific host when sending calls off node?

Sent from my iPhone

> On Jun 9, 2016, at 9:21 PM, Dave Horton <daveh at beachdognet.com> wrote:
> 
> I?ve seen a few recent posts on distributed presence work, and I am running into the same need.
> Like the other?s who have posted, I can?t use the same hostname across servers.
> In my case, this is because in order to make call transfer scenarios work I need
> to redirect calls sometimes to specific servers, and unique hostnames are required for that.
> 
> Is there another solution for this?  
> 
> If not, I?d be interested in working on a solution, as long as there was a view towards
> incorporating it in the mainstream code at some point (i.e., I wouldn't want to have to 
> keep patching releases going forward).
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From anthony.minessale at gmail.com  Fri Jun 10 09:16:06 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 10 Jun 2016 00:16:06 -0500
Subject: [Freeswitch-users] RFC 2833 payload type mismatch issue on
 bridged call
In-Reply-To: <B767942B-0074-4672-BCD3-4AD3DA5D9C99@beachdognet.com>
References: <B767942B-0074-4672-BCD3-4AD3DA5D9C99@beachdognet.com>
Message-ID: <CAKbxfG8uJ2tb4cAOQPoUH=rzGBHSHboUBk_StyYqBnV2NagEcg@mail.gmail.com>

https://freeswitch.org/jira


On Thursday, June 9, 2016, Dave Horton <daveh at beachdognet.com> wrote:

> I have a client sending an INVITE to FS with RFC 2833 offered as payload
> type 96.  My freeswitch device profile has "rfc2833-pt" set to 101 and
> "inbound-late-negotiation? set to true.
>
> The FS then bridges this call to a gateway, sending an INVITE offering pt
> 101 for 2833.
> The 200 OK on the B leg comes back (with pt 101), and FS then sends a 200
> OK to the original INVITE with pt 101.
> So, to summarize, the client has offered 101 for 2833 and the FS responds
> with an SDP with 96.
> At that point, DTMF does not work (DTMF generated by the client not
> recognized by the far end gateway).
>
> Is there a configuration setting that would tell FS to use the RDC 2833
> payload type from the A leg when bridging a call?
>
> Interestingly, I am only seeing this problem on servers I have upgraded
> (to  1.6.8) ? older servers do bridge the B leg offering pt 96 if that came
> in on the A leg.
> (Note, though, that the older servers are also not configured to do
> inbound-late-negotiation).
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <javascript:;>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From alhakeem at gmail.com  Fri Jun 10 09:35:13 2016
From: alhakeem at gmail.com (Abdul Hakeem)
Date: Fri, 10 Jun 2016 06:35:13 +0100
Subject: [Freeswitch-users] SHA-2 Implementation
Message-ID: <CAO=waFBp1ox46+Pe3hfsEw6gNWCCCiweC0P-9VEjrPRdMSWHCA@mail.gmail.com>

Hello,

Any plans of moving FS from SHA-1 to SHA-2 crypto hash support ?
I understand SHA-1 will be deprecated in Chrome by December, this will
cause all calls to Verto to fail.

Cheers,
AH
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From john.nash778 at gmail.com  Fri Jun 10 10:32:36 2016
From: john.nash778 at gmail.com (John Nash)
Date: Fri, 10 Jun 2016 12:02:36 +0530
Subject: [Freeswitch-users] Freeswitch changes payload type of codec
Message-ID: <CAA4rGPy7nBRKux0ZkrBTiXz8A72cg0U1zJ+Zct+ppjXaMSpDAw@mail.gmail.com>

I am facing same issue described in this old post
http://lists.freeswitch.org/pipermail/freeswitch-users/2009-June/042823.html

I am using media_bypass mode.

If any more information like traces etc needed I can provide.
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From deforceczt at gmail.com  Fri Jun 10 12:16:42 2016
From: deforceczt at gmail.com (Vladislav Ivanov)
Date: Fri, 10 Jun 2016 11:16:42 +0300
Subject: [Freeswitch-users] Safe SIP URI Formatting
Message-ID: <CAMCT=3TKjv7ShzkB+vDBqufhrFOb5BBXXQ=93+GD1qFahBrSng@mail.gmail.com>

Hello,

Is it possible to somehow alter the logic of Safe URI Formating to not
transform '#' sign to '%23'?
Maybe I can somehow escape the hash sign to not transform?
There is some equipments that cant do back-formating and reject calls with
"%23" signs with them.

Best Regards
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From flokrrr at gmail.com  Fri Jun 10 12:40:22 2016
From: flokrrr at gmail.com (Florent Krieg)
Date: Fri, 10 Jun 2016 10:40:22 +0200
Subject: [Freeswitch-users] Channel/session variables availability on
	execute_on_sip_reinvite
Message-ID: <CAABKsuRM0A-k5kMn+gsHHuQJYKVc6mT_mcqEdGDnt2MopL6t0g@mail.gmail.com>

Hello everybody,

My use case is pretty simple: I would like to react on SIP Re-INVITE
without sdp (empty body/content-length 0) so I have put an
execute_on_sip_reinvite instruction in my bridge string.

I have tried both to call the 'info' app and to call a custom shell system
script, passing the ${uuid} var and calling fs_cli uuid_dump $uuid, but
unfortunately it seems that when the SIP message reaches the application it
has been processed by the Sofia stack and I can't detect if there is an SDP
or not (using sdp variables, or any other actually).
Sometimes it gives me 0.0.0.0 as the remote_media_ip, sometimes it just
builds an SDP based on the previous one.
I have also tried to modify the re-INVITE by adding custom headers or
putting information in any
mandatory SIP header (like, say, Via), but these updates doesn't seem to be
available via any channel variable.

Is there any FS configuration parameter I could change to be able to access
this information?
Is there any other way I could do what I want?


Thanks in advance if you have any clue :)

Florent
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From mgg at giagnocavo.net  Fri Jun 10 12:41:24 2016
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Fri, 10 Jun 2016 08:41:24 +0000
Subject: [Freeswitch-users] Safe SIP URI Formatting
In-Reply-To: <CAMCT=3TKjv7ShzkB+vDBqufhrFOb5BBXXQ=93+GD1qFahBrSng@mail.gmail.com>
References: <CAMCT=3TKjv7ShzkB+vDBqufhrFOb5BBXXQ=93+GD1qFahBrSng@mail.gmail.com>
Message-ID: <BY2PR07MB6958A9C64E186B876C6B94ED4500@BY2PR07MB695.namprd07.prod.outlook.com>

This might be better suited to doing in a SIP proxy at the edge, like OpenSIPS. R-URI formatting is complicated. The # might be a URI fragment and be discarded by some UAs or might be illegal and rejected.

It might be worth it to try to avoid such URIs in the first place.

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Vladislav Ivanov
Sent: Friday, 10 June, 2016 2:17
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Safe SIP URI Formatting

Hello,

Is it possible to somehow alter the logic of Safe URI Formating to not transform '#' sign to '%23'?
Maybe I can somehow escape the hash sign to not transform?
There is some equipments that cant do back-formating and reject calls with "%23" signs with them.

Best Regards
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From royj at yandex.ru  Fri Jun 10 13:38:08 2016
From: royj at yandex.ru (royj at yandex.ru)
Date: Fri, 10 Jun 2016 12:38:08 +0300
Subject: [Freeswitch-users] Enterprise originate, SIP 302,
	sip_redirect_profile
Message-ID: <2489001465551488@web12o.yandex.ru>

Hi, all

The goal is call multiple destinations simultaneously. And when receiving SIP 302 from some of destination, the another calls must not be interrupted.
There is enterprise originate functionality for that. But we can not use  'sip_redirect_profile', 'sip_redirect_context' channel variables to route to redirection number in a dialplan when enterprise originate in action.
I mean there is no effect of setting that variables. When non enterprise originate (',' as separator)  'sip_redirect_profile', 'sip_redirect_context' channel variables work as expected, but calls to another destinations end.

Is there any way to use 'sip_redirect_profile', 'sip_redirect_context' in the enterprise originate?


From s.safarov at gmail.com  Fri Jun 10 15:34:04 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 10 Jun 2016 11:34:04 +0000
Subject: [Freeswitch-users] Safe SIP URI Formatting
In-Reply-To: <BY2PR07MB6958A9C64E186B876C6B94ED4500@BY2PR07MB695.namprd07.prod.outlook.com>
References: <CAMCT=3TKjv7ShzkB+vDBqufhrFOb5BBXXQ=93+GD1qFahBrSng@mail.gmail.com>
	<BY2PR07MB6958A9C64E186B876C6B94ED4500@BY2PR07MB695.namprd07.prod.outlook.com>
Message-ID: <CAHtxdDc1SR+8s7bzBW6B-Da8rxRtcj3chAbwr=XWjJQ=P4fS3g@mail.gmail.com>

Please check sip_invite_req_uri. May be it will work for you.

??, 10 ???. 2016 ?. ? 11:42, Michael Giagnocavo <mgg at giagnocavo.net>:

> This might be better suited to doing in a SIP proxy at the edge, like
> OpenSIPS. R-URI formatting is complicated. The # might be a URI fragment
> and be discarded by some UAs or might be illegal and rejected.
>
>
>
> It might be worth it to try to avoid such URIs in the first place.
>
>
>
> -Michael
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Vladislav
> Ivanov
> *Sent:* Friday, 10 June, 2016 2:17
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] Safe SIP URI Formatting
>
>
>
> Hello,
>
> Is it possible to somehow alter the logic of Safe URI Formating to not
> transform '#' sign to '%23'?
> Maybe I can somehow escape the hash sign to not transform?
> There is some equipments that cant do back-formating and reject calls with
> "%23" signs with them.
>
> Best Regards
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From s.safarov at gmail.com  Fri Jun 10 15:39:18 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 10 Jun 2016 11:39:18 +0000
Subject: [Freeswitch-users] Enterprise originate, SIP 302,
	sip_redirect_profile
In-Reply-To: <2489001465551488@web12o.yandex.ru>
References: <2489001465551488@web12o.yandex.ru>
Message-ID: <CAHtxdDexT2sWJVEC26zo4sHkXkWfXry1+SM0tM2iciUvfUT38A@mail.gmail.com>

You can try originate a call via loopback endpoint.

??, 10 ???. 2016 ?. ? 12:39, <royj at yandex.ru>:

> Hi, all
>
> The goal is call multiple destinations simultaneously. And when receiving
> SIP 302 from some of destination, the another calls must not be interrupted.
> There is enterprise originate functionality for that. But we can not use
> 'sip_redirect_profile', 'sip_redirect_context' channel variables to route
> to redirection number in a dialplan when enterprise originate in action.
> I mean there is no effect of setting that variables. When non enterprise
> originate (',' as separator)  'sip_redirect_profile',
> 'sip_redirect_context' channel variables work as expected, but calls to
> another destinations end.
>
> Is there any way to use 'sip_redirect_profile', 'sip_redirect_context' in
> the enterprise originate?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From deforceczt at gmail.com  Fri Jun 10 15:46:47 2016
From: deforceczt at gmail.com (Vladislav Ivanov)
Date: Fri, 10 Jun 2016 14:46:47 +0300
Subject: [Freeswitch-users] Safe SIP URI Formatting
In-Reply-To: <CAHtxdDc1SR+8s7bzBW6B-Da8rxRtcj3chAbwr=XWjJQ=P4fS3g@mail.gmail.com>
References: <CAMCT=3TKjv7ShzkB+vDBqufhrFOb5BBXXQ=93+GD1qFahBrSng@mail.gmail.com>
	<BY2PR07MB6958A9C64E186B876C6B94ED4500@BY2PR07MB695.namprd07.prod.outlook.com>
	<CAHtxdDc1SR+8s7bzBW6B-Da8rxRtcj3chAbwr=XWjJQ=P4fS3g@mail.gmail.com>
Message-ID: <CAMCT=3TX5vfWhqzg3pO+EEcaKpox9AaEdYcP+zComdR=C7Am0Q@mail.gmail.com>

Sergey,

Thank you, it worked flawlessly.

Michael,

Yeah, if the variable wouldn't work in Freeswitch I would've went with your
option.

Thank you all with helping resolving this issue.

2016-06-10 14:34 GMT+03:00 Sergey Safarov <s.safarov at gmail.com>:

> Please check sip_invite_req_uri. May be it will work for you.
>
> ??, 10 ???. 2016 ?. ? 11:42, Michael Giagnocavo <mgg at giagnocavo.net>:
>
>> This might be better suited to doing in a SIP proxy at the edge, like
>> OpenSIPS. R-URI formatting is complicated. The # might be a URI fragment
>> and be discarded by some UAs or might be illegal and rejected.
>>
>>
>>
>> It might be worth it to try to avoid such URIs in the first place.
>>
>>
>>
>> -Michael
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Vladislav
>> Ivanov
>> *Sent:* Friday, 10 June, 2016 2:17
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* [Freeswitch-users] Safe SIP URI Formatting
>>
>>
>>
>> Hello,
>>
>> Is it possible to somehow alter the logic of Safe URI Formating to not
>> transform '#' sign to '%23'?
>> Maybe I can somehow escape the hash sign to not transform?
>> There is some equipments that cant do back-formating and reject calls
>> with "%23" signs with them.
>>
>> Best Regards
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Fri Jun 10 15:56:45 2016
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 10 Jun 2016 12:56:45 +0100
Subject: [Freeswitch-users] System global mode variable
In-Reply-To: <57599135.50908@xmission.com>
References: <5756DF3E.8000004@xmission.com>
	<57599135.50908@xmission.com>
Message-ID: <CAFiqYu=t2792kgEUA7_sPN4nt=oeyacYGJUf_ipLh3+KNkiYbQ@mail.gmail.com>

Are you planning to change this at runtime? I seem to remember a post on
the mailing list from very long time ago (years) changing global variables
that were being read at the same time had a small risk of a race condition
that might segfault.

If so that has almost certainly been fixed by now, but if not you could
also look at using mod_hash or mod_db.

Your simple script can be done in xml by the way without lua using inline
api calls in the condition's field attributes. Entirely personal preference
which you use. Lua will fetch the variable just once so dies have that
advantage.





On Thursday, 9 June 2016, Pete Ashdown <pashdown at xmission.com> wrote:

> Answered my own question:
>
> In your dialplan:
> <action application="lua" data="script.lua"/>
>
> In script.lua:
> mode = freeswitch.getGlobalVariable("mode")
> if (mode == "open") then
>   session:speak("open")
> end
> if (mode == "techonly") then
>   session:speak("teck only")
> end
> if (mode == "event") then
>   session:speak("event")
> else
>   session:speak("i dont know")
> end
>
> Make sure the global is created somewhere, like vars.xml:
> <X-PRE-PROCESS cmd="set" data="mode=open"/>
>
>
> On 06/07/2016 08:50 AM, Pete Ashdown wrote:
> > What is the simplest way to have a conditional branch based on a global
> > variable in a dialplan?  Actually using a global variable and xml is
> > giving me mixed results.  Do I need to use Lua or another scripting
> > language?
> >
> > This is what I have, which doesn't give the results desired.   The
> > "speak" actions are merely debug placeholders.  I know there are better
> > ways to do conditional voice to text.
> >
> >       <condition regex="any" field="${cond(${mode} == 'open' ? YES :
> > NO)}" expression="^YES$">
> >             <action application="speak" data="open"/>
> >       </condition>
> >
> >       <condition regex="any" field="${cond(${mode} == 'tech only' ? YES
> > : NO)}" expression="^YES$">
> >             <action application="speak" data="teck only"/>
> >       </condition>
> >
> >       <condition regex="any" field="${cond(${mode} == 'event' ? YES :
> > NO)}" expression="^YES$">
> >             <action application="speak" data="event"/>
> >       </condition>
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <javascript:;>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From krice at freeswitch.org  Fri Jun 10 16:19:50 2016
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 10 Jun 2016 07:19:50 -0500
Subject: [Freeswitch-users] Freeswitch changes payload type of codec
In-Reply-To: <CAA4rGPy7nBRKux0ZkrBTiXz8A72cg0U1zJ+Zct+ppjXaMSpDAw@mail.gmail.com>
References: <CAA4rGPy7nBRKux0ZkrBTiXz8A72cg0U1zJ+Zct+ppjXaMSpDAw@mail.gmail.com>
Message-ID: <14ed01d1c312$63834530$2a89cf90$@freeswitch.org>

Buts should be filed at https://freeswitch.org/jira

 

Please Attach a complete log of a call doing this (see the note at the top of jira right under the menu bar on file attachments)

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of John Nash
Sent: Friday, June 10, 2016 1:33 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Freeswitch changes payload type of codec

 

I am facing same issue described in this old post http://lists.freeswitch.org/pipermail/freeswitch-users/2009-June/042823.html

 

I am using media_bypass mode.

 

If any more information like traces etc needed I can provide.

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From peter at olssononline.se  Fri Jun 10 16:41:46 2016
From: peter at olssononline.se (Peter Olsson)
Date: Fri, 10 Jun 2016 14:41:46 +0200
Subject: [Freeswitch-users] [ERR] helloworld.js:8 Exception:
 ReferenceError: File is not defined
In-Reply-To: <CAGVOjVXuXhdEbjrggx6aXmkADZ4Y1vXL0vEMaNLrwVfTdewdpg@mail.gmail.com>
References: <CAGVOjVXuXhdEbjrggx6aXmkADZ4Y1vXL0vEMaNLrwVfTdewdpg@mail.gmail.com>
Message-ID: <CAPEdYTidc-Fj4mrcTm-BKyjPoO7nLGbrE_tR=2k0jA17e1ZVNA@mail.gmail.com>

As referenced here, "File" is not available in mod_v8.

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

/Peter

2016-06-09 20:28 GMT+02:00 Aqs Younas <aqsyounas at gmail.com>:

> Greetings list.
>
> I am getting below error when i am trying open a file in mod_v8.
>
> 2016-06-02 01:40:44.908007 [ERR] helloworld.js:8 Exception:
> ReferenceError: File is not defined (near: "        fd = new
> File(baseFolder + "/" + menuId + ".json");")
>
> Any pointer is much appreciated
>
> Thanks.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Fri Jun 10 17:49:02 2016
From: brian at freeswitch.org (Brian West)
Date: Fri, 10 Jun 2016 08:49:02 -0500
Subject: [Freeswitch-users] Safe SIP URI Formatting
In-Reply-To: <CAHtxdDc1SR+8s7bzBW6B-Da8rxRtcj3chAbwr=XWjJQ=P4fS3g@mail.gmail.com>
References: <CAMCT=3TKjv7ShzkB+vDBqufhrFOb5BBXXQ=93+GD1qFahBrSng@mail.gmail.com>
	<BY2PR07MB6958A9C64E186B876C6B94ED4500@BY2PR07MB695.namprd07.prod.outlook.com>
	<CAHtxdDc1SR+8s7bzBW6B-Da8rxRtcj3chAbwr=XWjJQ=P4fS3g@mail.gmail.com>
Message-ID: <CAEJMVkBjkQdsh-i0OySZ8kwN4wkmbGN9VBX3GruEPboD33yDSQ@mail.gmail.com>

If your sending a # in the user part of the URI its invalid per the RFC, We
default to the RFC behavior, you can disable it.
 sofia_suppress_url_encoding=true in your dial string.

/b

On Fri, Jun 10, 2016 at 6:34 AM, Sergey Safarov <s.safarov at gmail.com> wrote:

> Please check sip_invite_req_uri. May be it will work for you.
>
> ??, 10 ???. 2016 ?. ? 11:42, Michael Giagnocavo <mgg at giagnocavo.net>:
>
>> This might be better suited to doing in a SIP proxy at the edge, like
>> OpenSIPS. R-URI formatting is complicated. The # might be a URI fragment
>> and be discarded by some UAs or might be illegal and rejected.
>>
>>
>>
>> It might be worth it to try to avoid such URIs in the first place.
>>
>>
>>
>> -Michael
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Vladislav
>> Ivanov
>> *Sent:* Friday, 10 June, 2016 2:17
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* [Freeswitch-users] Safe SIP URI Formatting
>>
>>
>>
>> Hello,
>>
>> Is it possible to somehow alter the logic of Safe URI Formating to not
>> transform '#' sign to '%23'?
>> Maybe I can somehow escape the hash sign to not transform?
>> There is some equipments that cant do back-formating and reject calls
>> with "%23" signs with them.
>>
>> Best Regards
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com
https://www.gofundme.com/freeswitch_ubuntu

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

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From brian at freeswitch.org  Fri Jun 10 17:50:01 2016
From: brian at freeswitch.org (Brian West)
Date: Fri, 10 Jun 2016 08:50:01 -0500
Subject: [Freeswitch-users] Enterprise originate, SIP 302,
	sip_redirect_profile
In-Reply-To: <CAHtxdDexT2sWJVEC26zo4sHkXkWfXry1+SM0tM2iciUvfUT38A@mail.gmail.com>
References: <2489001465551488@web12o.yandex.ru>
	<CAHtxdDexT2sWJVEC26zo4sHkXkWfXry1+SM0tM2iciUvfUT38A@mail.gmail.com>
Message-ID: <CAEJMVkCAP9+G53qhOx5fp0nwxDFbO6RgEfstpkf-ee5bSZgqMg@mail.gmail.com>

Nope,

Set the variable outbound_redirect_fatal=true, This will cause the 302's in
a group dial to just fall away and not be followed.

/b


On Fri, Jun 10, 2016 at 6:39 AM, Sergey Safarov <s.safarov at gmail.com> wrote:

> You can try originate a call via loopback endpoint.
>
> ??, 10 ???. 2016 ?. ? 12:39, <royj at yandex.ru>:
>
>> Hi, all
>>
>> The goal is call multiple destinations simultaneously. And when receiving
>> SIP 302 from some of destination, the another calls must not be interrupted.
>> There is enterprise originate functionality for that. But we can not use
>> 'sip_redirect_profile', 'sip_redirect_context' channel variables to route
>> to redirection number in a dialplan when enterprise originate in action.
>> I mean there is no effect of setting that variables. When non enterprise
>> originate (',' as separator)  'sip_redirect_profile',
>> 'sip_redirect_context' channel variables work as expected, but calls to
>> another destinations end.
>>
>> Is there any way to use 'sip_redirect_profile', 'sip_redirect_context' in
>> the enterprise originate?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com
https://www.gofundme.com/freeswitch_ubuntu

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

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From deforceczt at gmail.com  Fri Jun 10 17:58:20 2016
From: deforceczt at gmail.com (Vladislav Ivanov)
Date: Fri, 10 Jun 2016 16:58:20 +0300
Subject: [Freeswitch-users] Safe SIP URI Formatting
In-Reply-To: <CAEJMVkBjkQdsh-i0OySZ8kwN4wkmbGN9VBX3GruEPboD33yDSQ@mail.gmail.com>
References: <CAMCT=3TKjv7ShzkB+vDBqufhrFOb5BBXXQ=93+GD1qFahBrSng@mail.gmail.com>
	<BY2PR07MB6958A9C64E186B876C6B94ED4500@BY2PR07MB695.namprd07.prod.outlook.com>
	<CAHtxdDc1SR+8s7bzBW6B-Da8rxRtcj3chAbwr=XWjJQ=P4fS3g@mail.gmail.com>
	<CAEJMVkBjkQdsh-i0OySZ8kwN4wkmbGN9VBX3GruEPboD33yDSQ@mail.gmail.com>
Message-ID: <CAMCT=3TJLVfixCUHAZHpc3uzPDhTf_XB7GqE3zxk6N8acKeZMA@mail.gmail.com>

I understand that is not valid by RFC.
It's about vendors equipment wont solve the "compliance by RFC".

I'd gladly stay RFC, but with this vendor it's sadly not possible.

2016-06-10 16:49 GMT+03:00 Brian West <brian at freeswitch.org>:

> If your sending a # in the user part of the URI its invalid per the RFC,
> We default to the RFC behavior, you can disable it.
>  sofia_suppress_url_encoding=true in your dial string.
>
> /b
>
> On Fri, Jun 10, 2016 at 6:34 AM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
>
>> Please check sip_invite_req_uri. May be it will work for you.
>>
>> ??, 10 ???. 2016 ?. ? 11:42, Michael Giagnocavo <mgg at giagnocavo.net>:
>>
>>> This might be better suited to doing in a SIP proxy at the edge, like
>>> OpenSIPS. R-URI formatting is complicated. The # might be a URI fragment
>>> and be discarded by some UAs or might be illegal and rejected.
>>>
>>>
>>>
>>> It might be worth it to try to avoid such URIs in the first place.
>>>
>>>
>>>
>>> -Michael
>>>
>>>
>>>
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Vladislav
>>> Ivanov
>>> *Sent:* Friday, 10 June, 2016 2:17
>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> *Subject:* [Freeswitch-users] Safe SIP URI Formatting
>>>
>>>
>>>
>>> Hello,
>>>
>>> Is it possible to somehow alter the logic of Safe URI Formating to not
>>> transform '#' sign to '%23'?
>>> Maybe I can somehow escape the hash sign to not transform?
>>> There is some equipments that cant do back-formating and reject calls
>>> with "%23" signs with them.
>>>
>>> Best Regards
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
> https://www.gofundme.com/freeswitch_ubuntu
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From daveh at beachdognet.com  Fri Jun 10 18:59:36 2016
From: daveh at beachdognet.com (Dave Horton)
Date: Fri, 10 Jun 2016 10:59:36 -0400
Subject: [Freeswitch-users] RFC 2833 payload type mismatch issue on
	bridged call
In-Reply-To: <CAKbxfG8uJ2tb4cAOQPoUH=rzGBHSHboUBk_StyYqBnV2NagEcg@mail.gmail.com>
References: <B767942B-0074-4672-BCD3-4AD3DA5D9C99@beachdognet.com>
	<CAKbxfG8uJ2tb4cAOQPoUH=rzGBHSHboUBk_StyYqBnV2NagEcg@mail.gmail.com>
Message-ID: <CDCDA0B0-D979-4C0A-8A7A-F24150D35742@beachdognet.com>

https://freeswitch.org/jira/browse/FS-9252 <https://freeswitch.org/jira/browse/FS-9252>

On Jun 10, 2016, at 1:16 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

https://freeswitch.org/jira <https://freeswitch.org/jira>


On Thursday, June 9, 2016, Dave Horton <daveh at beachdognet.com <mailto:daveh at beachdognet.com>> wrote:
I have a client sending an INVITE to FS with RFC 2833 offered as payload type 96.  My freeswitch device profile has "rfc2833-pt" set to 101 and "inbound-late-negotiation? set to true.

The FS then bridges this call to a gateway, sending an INVITE offering pt 101 for 2833.
The 200 OK on the B leg comes back (with pt 101), and FS then sends a 200 OK to the original INVITE with pt 101.
So, to summarize, the client has offered 101 for 2833 and the FS responds with an SDP with 96.
At that point, DTMF does not work (DTMF generated by the client not recognized by the far end gateway).

Is there a configuration setting that would tell FS to use the RDC 2833 payload type from the A leg when bridging a call?

Interestingly, I am only seeing this problem on servers I have upgraded (to  1.6.8) ? older servers do bridge the B leg offering pt 96 if that came in on the A leg.
(Note, though, that the older servers are also not configured to do inbound-late-negotiation).
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <javascript:;>
http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>

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

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

-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

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

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

https://www.youtube.com/watch?v=9XXgW34t40s <https://www.youtube.com/watch?v=9XXgW34t40s>
https://www.youtube.com/watch?v=NLaDpGQuZDA <https://www.youtube.com/watch?v=NLaDpGQuZDA>

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

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

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

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From stefan.mititelu92 at gmail.com  Fri Jun 10 14:05:22 2016
From: stefan.mititelu92 at gmail.com (Mititelu Stefan)
Date: Fri, 10 Jun 2016 13:05:22 +0300
Subject: [Freeswitch-users] Dynamic Group Call
Message-ID: <CAL82Oa1YksiVpFFxovF8ghy_3XE_Rkp45wisj9Zt4Mnp0jMnnA@mail.gmail.com>

Hi,

I'm a Freeswitch newbee and I'm trying to set up dynamic group calls.

Freeswitch will receive a header with "Group-Users:user1,user2.." and what
I want to do is to "transfer" to a dinamycally created context that will
look like:
      <action application="conference_set_auto_outcall" data="user1"/>
      <action application="conference_set_auto_outcall" data="user2"/>

Is this possible? Can you give me some references?

>From what I see in some examples one can transfer only to existent XML
context:
    <action application="transfer" data="1105 XML default"/>

Thanks,
Stefan
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From fan_x07 at yahoo.com  Fri Jun 10 14:18:51 2016
From: fan_x07 at yahoo.com (FAN_X FAN_X)
Date: Fri, 10 Jun 2016 10:18:51 +0000 (UTC)
Subject: [Freeswitch-users] Dynamic Group Call
References: <1165532784.799074.1465553931212.JavaMail.yahoo.ref@mail.yahoo.com>
Message-ID: <1165532784.799074.1465553931212.JavaMail.yahoo@mail.yahoo.com>

Hi,

I'm a Freeswitch newbee and I'm trying to set up dynamic group calls.

Freeswitch will receive a header with "Group-Users:user1,user2.." and what I want to do is to "transfer" to a dinamycally created context that will look like:
????? <action application="conference_set_auto_outcall" data="user1"/>
????? <action application="conference_set_auto_outcall" data="user2"/>

Is this possible? Can you give me some references?

>From what I see in some examples one can transfer only to existent XML context:
??? <action application="transfer" data="1105 XML default"/>

Thanks,
Stefan
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From ralph at wesleycloversolutions.com  Fri Jun 10 19:56:54 2016
From: ralph at wesleycloversolutions.com (Ralph Willett)
Date: Fri, 10 Jun 2016 15:56:54 +0000
Subject: [Freeswitch-users] Activating MWI from a Mitel 3300
Message-ID: <CY1PR10MB094931DFFAAEE06145A1A781A8500@CY1PR10MB0949.namprd10.prod.outlook.com>

I have a freeswitch connecting to a Mitel 3300 via sip trunks.
We are using the Mitel embedded voice mail.
I'm trying to figure out how to light the MWI light on sets running off of the freeswitch.
Is it possible to light a MWI across the SIP trunks?
Does anyone have suggestions on how to proceed to figure out how to do this?

Dryaquaman
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From mike at jerris.com  Fri Jun 10 21:14:37 2016
From: mike at jerris.com (Michael Jerris)
Date: Fri, 10 Jun 2016 13:14:37 -0400
Subject: [Freeswitch-users] Channel/session variables availability on
	execute_on_sip_reinvite
In-Reply-To: <CAABKsuRM0A-k5kMn+gsHHuQJYKVc6mT_mcqEdGDnt2MopL6t0g@mail.gmail.com>
References: <CAABKsuRM0A-k5kMn+gsHHuQJYKVc6mT_mcqEdGDnt2MopL6t0g@mail.gmail.com>
Message-ID: <9CA4C4E8-9FE0-4413-BA60-7A303EF4AFA6@jerris.com>

I don't think we have the sdp parsed and processed at the point we get that callback.  Check the variables to see if the raw sdp is available in the var yet or not.

> On Jun 10, 2016, at 4:40 AM, Florent Krieg <flokrrr at gmail.com> wrote:
> 
> Hello everybody,
> 
> My use case is pretty simple: I would like to react on SIP Re-INVITE without sdp (empty body/content-length 0) so I have put an execute_on_sip_reinvite instruction in my bridge string.
> 
> I have tried both to call the 'info' app and to call a custom shell system script, passing the ${uuid} var and calling fs_cli uuid_dump $uuid, but unfortunately it seems that when the SIP message reaches the application it has been processed by the Sofia stack and I can't detect if there is an SDP or not (using sdp variables, or any other actually).
> Sometimes it gives me 0.0.0.0 as the remote_media_ip, sometimes it just builds an SDP based on the previous one.
> I have also tried to modify the re-INVITE by adding custom headers or putting information in any
> mandatory SIP header (like, say, Via), but these updates doesn't seem to be available via any channel variable.
> 
> Is there any FS configuration parameter I could change to be able to access this information?
> Is there any other way I could do what I want?
> 
> 
> Thanks in advance if you have any clue :)
> 
> Florent



From mike at jerris.com  Fri Jun 10 21:19:02 2016
From: mike at jerris.com (Michael Jerris)
Date: Fri, 10 Jun 2016 13:19:02 -0400
Subject: [Freeswitch-users] Dynamic Group Call
In-Reply-To: <CAL82Oa1YksiVpFFxovF8ghy_3XE_Rkp45wisj9Zt4Mnp0jMnnA@mail.gmail.com>
References: <CAL82Oa1YksiVpFFxovF8ghy_3XE_Rkp45wisj9Zt4Mnp0jMnnA@mail.gmail.com>
Message-ID: <12523881-DD17-4269-A9D9-CB7CF28B18C1@jerris.com>

not easy to do in xml dialplan but probably doable in any of the dynamic dialplan bindings via xml_curl or one of the embedded langs like lua

> On Jun 10, 2016, at 6:05 AM, Mititelu Stefan <stefan.mititelu92 at gmail.com> wrote:
> 
> Hi,
> 
> I'm a Freeswitch newbee and I'm trying to set up dynamic group calls.
> 
> Freeswitch will receive a header with "Group-Users:user1,user2.." and what I want to do is to "transfer" to a dinamycally created context that will look like:
>       <action application="conference_set_auto_outcall" data="user1"/>
>       <action application="conference_set_auto_outcall" data="user2"/>
> 
> Is this possible? Can you give me some references?
> 
> From what I see in some examples one can transfer only to existent XML context:
>     <action application="transfer" data="1105 XML default"/>
> 
> Thanks,
> Stefan

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From ba.lerest at gmail.com  Fri Jun 10 21:28:45 2016
From: ba.lerest at gmail.com (Bastien LE REST)
Date: Fri, 10 Jun 2016 18:28:45 +0100
Subject: [Freeswitch-users] ICE issue
In-Reply-To: <CAL+i1s-_Sg8iN+ygKS0yVhM=_cdvvtC29at3L=5hHzcWyE3f8Q@mail.gmail.com>
References: <CAL+i1s-_Sg8iN+ygKS0yVhM=_cdvvtC29at3L=5hHzcWyE3f8Q@mail.gmail.com>
Message-ID: <CAL+i1s8at0j19ZjUnYVve_NDLaHLh5Gzbze=KYaw5v4j7_VYTA@mail.gmail.com>

Bonjour,

I thought the issue was solved since the calls work just fine now with
PJSIP on iOS, however the calls don't work with PJSIP on Android and I
don't understand why, the PJSIP configuration is exactly the same on both
devices.

FS finds a candidate for RTP but it doesn't find a candidate for RTCP.
Please note that I only have 3 candidate.

FS trace: https://pastebin.freeswitch.org/view/8d4df983

Could you please have a look at the trace, and let me know what am I doing
wrong?

Thank you very much in advance for your help in this matter.

Kind regards,

Bastien.

On 4 June 2016 at 13:44, Bastien LE REST <ba.lerest at gmail.com> wrote:

> Hi,
>
> When I enable ICE using PJSIP, FreeSwitch answers "488 Not Acceptable
> Here". When I disable ICE, I don't have any problems, the calls work just
> fine. I tried to use another SIP client with ICE and the call works fine.
> So I think the problem has to do with PJSIP.
>
> I'm working on the migration from an Asterisk architecture to a FreeSwitch
> architecture. I have to deal with PJSIP and ICE support because my client
> is a mobile application who is already published on the stores.
>
> PJSIP trace : https://pastebin.freeswitch.org/view/ddeaae4b
> FreeSwitch trace: https://pastebin.freeswitch.org/view/d380e01d
>
> Could you please have a look at the trace and tell me what am I doing
> wrong? Thank you in advance.
>
> Kind regards,
> Bastien.
>
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From vittico at gmail.com  Fri Jun 10 22:26:46 2016
From: vittico at gmail.com (Victor Medina)
Date: Fri, 10 Jun 2016 14:26:46 -0400
Subject: [Freeswitch-users] ICE issue
In-Reply-To: <CAL+i1s8at0j19ZjUnYVve_NDLaHLh5Gzbze=KYaw5v4j7_VYTA@mail.gmail.com>
References: <CAL+i1s-_Sg8iN+ygKS0yVhM=_cdvvtC29at3L=5hHzcWyE3f8Q@mail.gmail.com>
	<CAL+i1s8at0j19ZjUnYVve_NDLaHLh5Gzbze=KYaw5v4j7_VYTA@mail.gmail.com>
Message-ID: <CAEA-bfHwFWopNNvGpYj-8oOLe9EBJ-UA4LQDX73w5fkZYx-6Gg@mail.gmail.com>

There is seems to be a codec issue.
Are you sure codecs are right on the Android side?


Sin mas a que hacer referencia,

Victor Medina

On Fri, Jun 10, 2016 at 1:28 PM, Bastien LE REST <ba.lerest at gmail.com>
wrote:

> Bonjour,
>
> I thought the issue was solved since the calls work just fine now with
> PJSIP on iOS, however the calls don't work with PJSIP on Android and I
> don't understand why, the PJSIP configuration is exactly the same on both
> devices.
>
> FS finds a candidate for RTP but it doesn't find a candidate for RTCP.
> Please note that I only have 3 candidate.
>
> FS trace: https://pastebin.freeswitch.org/view/8d4df983
>
> Could you please have a look at the trace, and let me know what am I doing
> wrong?
>
> Thank you very much in advance for your help in this matter.
>
> Kind regards,
>
> Bastien.
>
> On 4 June 2016 at 13:44, Bastien LE REST <ba.lerest at gmail.com> wrote:
>
>> Hi,
>>
>> When I enable ICE using PJSIP, FreeSwitch answers "488 Not Acceptable
>> Here". When I disable ICE, I don't have any problems, the calls work just
>> fine. I tried to use another SIP client with ICE and the call works fine.
>> So I think the problem has to do with PJSIP.
>>
>> I'm working on the migration from an Asterisk architecture to a
>> FreeSwitch architecture. I have to deal with PJSIP and ICE support because
>> my client is a mobile application who is already published on the stores.
>>
>> PJSIP trace : https://pastebin.freeswitch.org/view/ddeaae4b
>> FreeSwitch trace: https://pastebin.freeswitch.org/view/d380e01d
>>
>> Could you please have a look at the trace and tell me what am I doing
>> wrong? Thank you in advance.
>>
>> Kind regards,
>> Bastien.
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vittico at gmail.com  Fri Jun 10 22:33:09 2016
From: vittico at gmail.com (Victor Medina)
Date: Fri, 10 Jun 2016 14:33:09 -0400
Subject: [Freeswitch-users] ICE issue
In-Reply-To: <CAEA-bfHwFWopNNvGpYj-8oOLe9EBJ-UA4LQDX73w5fkZYx-6Gg@mail.gmail.com>
References: <CAL+i1s-_Sg8iN+ygKS0yVhM=_cdvvtC29at3L=5hHzcWyE3f8Q@mail.gmail.com>
	<CAL+i1s8at0j19ZjUnYVve_NDLaHLh5Gzbze=KYaw5v4j7_VYTA@mail.gmail.com>
	<CAEA-bfHwFWopNNvGpYj-8oOLe9EBJ-UA4LQDX73w5fkZYx-6Gg@mail.gmail.com>
Message-ID: <CAEA-bfGHCH51VZbODgTeEX5kGd1n+ZRociW9d76gwGck1UXqHw@mail.gmail.com>

More...

On iOS you seem to be sending:


   1.    a=rtpmap:98 speex/16000
   2.    a=rtpmap:97 speex/8000
   3.    a=rtpmap:99 speex/32000
   4.    a=rtpmap:104 iLBC/8000
   5.    a=fmtp:104 mode=30
   6.    a=rtpmap:3 GSM/8000
   7.    a=rtpmap:0 PCMU/8000
   8.    a=rtpmap:8 PCMA/8000
   9.    a=rtpmap:9 G722/8000


While on Android I can only see:


   1.    a=rtpmap:98 speex/16000
   2.    a=rtpmap:97 speex/8000
   3.    a=rtpmap:99 speex/32000
   4.    a=rtpmap:104 iLBC/8000


FS is correctly complaining about it:

2016-06-10 12:48:33.533730 [ERR] mod_sofia.c:2346 CODEC NEGOTIATION ERROR.
SDP:


Sin mas a que hacer referencia,

Victor Medina

On Fri, Jun 10, 2016 at 2:26 PM, Victor Medina <vittico at gmail.com> wrote:

> There is seems to be a codec issue.
> Are you sure codecs are right on the Android side?
>
>
> Sin mas a que hacer referencia,
>
> Victor Medina
>
> On Fri, Jun 10, 2016 at 1:28 PM, Bastien LE REST <ba.lerest at gmail.com>
> wrote:
>
>> Bonjour,
>>
>> I thought the issue was solved since the calls work just fine now with
>> PJSIP on iOS, however the calls don't work with PJSIP on Android and I
>> don't understand why, the PJSIP configuration is exactly the same on both
>> devices.
>>
>> FS finds a candidate for RTP but it doesn't find a candidate for RTCP.
>> Please note that I only have 3 candidate.
>>
>> FS trace: https://pastebin.freeswitch.org/view/8d4df983
>>
>> Could you please have a look at the trace, and let me know what am I
>> doing wrong?
>>
>> Thank you very much in advance for your help in this matter.
>>
>> Kind regards,
>>
>> Bastien.
>>
>> On 4 June 2016 at 13:44, Bastien LE REST <ba.lerest at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> When I enable ICE using PJSIP, FreeSwitch answers "488 Not Acceptable
>>> Here". When I disable ICE, I don't have any problems, the calls work just
>>> fine. I tried to use another SIP client with ICE and the call works fine.
>>> So I think the problem has to do with PJSIP.
>>>
>>> I'm working on the migration from an Asterisk architecture to a
>>> FreeSwitch architecture. I have to deal with PJSIP and ICE support because
>>> my client is a mobile application who is already published on the stores.
>>>
>>> PJSIP trace : https://pastebin.freeswitch.org/view/ddeaae4b
>>> FreeSwitch trace: https://pastebin.freeswitch.org/view/d380e01d
>>>
>>> Could you please have a look at the trace and tell me what am I doing
>>> wrong? Thank you in advance.
>>>
>>> Kind regards,
>>> Bastien.
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From vittico at gmail.com  Fri Jun 10 22:45:01 2016
From: vittico at gmail.com (Victor Medina)
Date: Fri, 10 Jun 2016 14:45:01 -0400
Subject: [Freeswitch-users] SHA-2 Implementation
In-Reply-To: <CAO=waFBp1ox46+Pe3hfsEw6gNWCCCiweC0P-9VEjrPRdMSWHCA@mail.gmail.com>
References: <CAO=waFBp1ox46+Pe3hfsEw6gNWCCCiweC0P-9VEjrPRdMSWHCA@mail.gmail.com>
Message-ID: <CAEA-bfGVCV2Zk8MMHdY0JAtzaqU1ibnov2kvaZJfiEXqb1=BUA@mail.gmail.com>

Abdul.

As far as I know this is already implemented:

m=audio 26716 UDP/TLS/RTP/SAVPF 111 126
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=0; minptime=10; stereo=1; sprop-stereo=1
a=rtpmap:126 telephone-event/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=fingerprint:sha-256
5D:36:EC:52:16:CF:18:F8:79:CE:9A:BF:7B:1D:F1:08:3C:B9:52:50:91:1F:CE:13:E2:94:18:F5:55:C2:15:1B




Sin mas a que hacer referencia,

Victor Medina

On Fri, Jun 10, 2016 at 1:35 AM, Abdul Hakeem <alhakeem at gmail.com> wrote:

> Hello,
>
> Any plans of moving FS from SHA-1 to SHA-2 crypto hash support ?
> I understand SHA-1 will be deprecated in Chrome by December, this will
> cause all calls to Verto to fail.
>
> Cheers,
> AH
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kbdfck at gmail.com  Fri Jun 10 22:57:12 2016
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Fri, 10 Jun 2016 21:57:12 +0300
Subject: [Freeswitch-users] Enterprise originate, SIP 302,
	sip_redirect_profile
In-Reply-To: <CAEJMVkCAP9+G53qhOx5fp0nwxDFbO6RgEfstpkf-ee5bSZgqMg@mail.gmail.com>
References: <2489001465551488@web12o.yandex.ru>
	<CAHtxdDexT2sWJVEC26zo4sHkXkWfXry1+SM0tM2iciUvfUT38A@mail.gmail.com>
	<CAEJMVkCAP9+G53qhOx5fp0nwxDFbO6RgEfstpkf-ee5bSZgqMg@mail.gmail.com>
Message-ID: <CA+GmxbbG45a3O7+CYxtoerN386bJnwz6FEF9nUWFvZkpd_1pGQ@mail.gmail.com>

It will stop handling of 302 completely, right?
Could that 302 be followed manually for some leg of call? Can we processed
it in ESL or dialplan?

2016-06-10 16:50 GMT+03:00 Brian West <brian at freeswitch.org>:

> Nope,
>
> Set the variable outbound_redirect_fatal=true, This will cause the 302's
> in a group dial to just fall away and not be followed.
>
> /b
>
>
> On Fri, Jun 10, 2016 at 6:39 AM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
>
>> You can try originate a call via loopback endpoint.
>>
>> ??, 10 ???. 2016 ?. ? 12:39, <royj at yandex.ru>:
>>
>>> Hi, all
>>>
>>> The goal is call multiple destinations simultaneously. And when
>>> receiving SIP 302 from some of destination, the another calls must not be
>>> interrupted.
>>> There is enterprise originate functionality for that. But we can not
>>> use  'sip_redirect_profile', 'sip_redirect_context' channel variables to
>>> route to redirection number in a dialplan when enterprise originate in
>>> action.
>>> I mean there is no effect of setting that variables. When non enterprise
>>> originate (',' as separator)  'sip_redirect_profile',
>>> 'sip_redirect_context' channel variables work as expected, but calls to
>>> another destinations end.
>>>
>>> Is there any way to use 'sip_redirect_profile', 'sip_redirect_context'
>>> in the enterprise originate?
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
> https://www.gofundme.com/freeswitch_ubuntu
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From lexxua at gmail.com  Sat Jun 11 00:38:41 2016
From: lexxua at gmail.com (Volodymyr Fedorov)
Date: Fri, 10 Jun 2016 22:38:41 +0200
Subject: [Freeswitch-users] Enable nat-options-ping per user
Message-ID: <CAO_GGApncN4hN_qnxWSBOg6MbKTkoCcS0j5YKvThrCp+hW0OYA@mail.gmail.com>

Hi Community,
I have a question:
according to wiki:
https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal
parameter
<param name="nat-options-ping" value="true"/> may be assigned to profile.
But may it be assigned to particular user ?
I tried to do this like that:
  <user id="pbxuser">
    <params>
      <param name="password" value="PwgEn123"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="1000"/>
      <variable name="user_context" value="mypbx"/>
      <variable name="callgroup" value="techsupport"/>
        <variable name="sip-force-expires" value="180"/>
        <variable name="sip-force-contact"
value="NDLB-connectile-dysfunction"/>
        <variable name="nat-options-ping" value="true"/>
</variables>
  </user>

But freeswitch does not send option to registered endpoint.

Thanks !

-- 
Best regards,
Volodymyr
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From colin.morelli at gmail.com  Sat Jun 11 00:38:34 2016
From: colin.morelli at gmail.com (Colin Morelli)
Date: Fri, 10 Jun 2016 20:38:34 +0000
Subject: [Freeswitch-users] play_and_get_digits issue
Message-ID: <CAPtU-UoiTHgrEoL_pAdQT9GDjsRZLbt5V-MXGCW8+By0XhQwMA@mail.gmail.com>

Hello,

I'm trying to use play_and_get_digits with mod_ssml to speak an SSML string
while listening for DTMF tones. However, I seem to be having an issue with
how to escape the playback string. If I execute:

<action application="play_and_get_digits" data="0 1 1 5000 #
silence_stream://5000 silence_stream://250 digits" />

Everything works great, but if I instead do

<action application="play_and_get_digits" data="0 1 1 5000 #
ssml://<speak>Hello there</speak> silence_stream://250 digits" />

I hear the audio and FS collects the digits, but it doesn't assign the
dialplan variable. I can only assume that this is a result of the space in
the "Hello there" string, but I can't figure out how to properly escape it
(wrapping in single quotes doesn't seem to work, setting a channel variable
and using that doesn't seem to work).

Am I missing something obvious here? Tried to search the documentation for
escaping dialplan variables but didn't find anything.

Thanks,
Colin
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From ba.lerest at gmail.com  Sat Jun 11 13:36:24 2016
From: ba.lerest at gmail.com (Bastien LE REST)
Date: Sat, 11 Jun 2016 10:36:24 +0100
Subject: [Freeswitch-users] ICE issue
In-Reply-To: <CAEA-bfGHCH51VZbODgTeEX5kGd1n+ZRociW9d76gwGck1UXqHw@mail.gmail.com>
References: <CAL+i1s-_Sg8iN+ygKS0yVhM=_cdvvtC29at3L=5hHzcWyE3f8Q@mail.gmail.com>
	<CAL+i1s8at0j19ZjUnYVve_NDLaHLh5Gzbze=KYaw5v4j7_VYTA@mail.gmail.com>
	<CAEA-bfHwFWopNNvGpYj-8oOLe9EBJ-UA4LQDX73w5fkZYx-6Gg@mail.gmail.com>
	<CAEA-bfGHCH51VZbODgTeEX5kGd1n+ZRociW9d76gwGck1UXqHw@mail.gmail.com>
Message-ID: <CAL+i1s9V6ZYJYb64VNdwjgB0z83He1a7Rk8uaqnESL3_SVuiUg@mail.gmail.com>

Hello Victor,

Thank you for your reply. I tried to add both iLBC and Speex as a priority
codec in the codecs list but unfortunately that didn't fix the problem.

Is it normal that FS doesn't find a candidate for RTCP?

Kind regards,
Bastien.

On 10 June 2016 at 19:33, Victor Medina <vittico at gmail.com> wrote:

> More...
>
> On iOS you seem to be sending:
>
>
>    1.    a=rtpmap:98 speex/16000
>    2.    a=rtpmap:97 speex/8000
>    3.    a=rtpmap:99 speex/32000
>    4.    a=rtpmap:104 iLBC/8000
>    5.    a=fmtp:104 mode=30
>    6.    a=rtpmap:3 GSM/8000
>    7.    a=rtpmap:0 PCMU/8000
>    8.    a=rtpmap:8 PCMA/8000
>    9.    a=rtpmap:9 G722/8000
>
>
> While on Android I can only see:
>
>
>    1.    a=rtpmap:98 speex/16000
>    2.    a=rtpmap:97 speex/8000
>    3.    a=rtpmap:99 speex/32000
>    4.    a=rtpmap:104 iLBC/8000
>
>
> FS is correctly complaining about it:
>
> 2016-06-10 12:48:33.533730 [ERR] mod_sofia.c:2346 CODEC NEGOTIATION
> ERROR.  SDP:
>
>
> Sin mas a que hacer referencia,
>
> Victor Medina
>
> On Fri, Jun 10, 2016 at 2:26 PM, Victor Medina <vittico at gmail.com> wrote:
>
>> There is seems to be a codec issue.
>> Are you sure codecs are right on the Android side?
>>
>>
>> Sin mas a que hacer referencia,
>>
>> Victor Medina
>>
>> On Fri, Jun 10, 2016 at 1:28 PM, Bastien LE REST <ba.lerest at gmail.com>
>> wrote:
>>
>>> Bonjour,
>>>
>>> I thought the issue was solved since the calls work just fine now with
>>> PJSIP on iOS, however the calls don't work with PJSIP on Android and I
>>> don't understand why, the PJSIP configuration is exactly the same on both
>>> devices.
>>>
>>> FS finds a candidate for RTP but it doesn't find a candidate for RTCP.
>>> Please note that I only have 3 candidate.
>>>
>>> FS trace: https://pastebin.freeswitch.org/view/8d4df983
>>>
>>> Could you please have a look at the trace, and let me know what am I
>>> doing wrong?
>>>
>>> Thank you very much in advance for your help in this matter.
>>>
>>> Kind regards,
>>>
>>> Bastien.
>>>
>>> On 4 June 2016 at 13:44, Bastien LE REST <ba.lerest at gmail.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> When I enable ICE using PJSIP, FreeSwitch answers "488 Not Acceptable
>>>> Here". When I disable ICE, I don't have any problems, the calls work just
>>>> fine. I tried to use another SIP client with ICE and the call works fine.
>>>> So I think the problem has to do with PJSIP.
>>>>
>>>> I'm working on the migration from an Asterisk architecture to a
>>>> FreeSwitch architecture. I have to deal with PJSIP and ICE support because
>>>> my client is a mobile application who is already published on the stores.
>>>>
>>>> PJSIP trace : https://pastebin.freeswitch.org/view/ddeaae4b
>>>> FreeSwitch trace: https://pastebin.freeswitch.org/view/d380e01d
>>>>
>>>> Could you please have a look at the trace and tell me what am I doing
>>>> wrong? Thank you in advance.
>>>>
>>>> Kind regards,
>>>> Bastien.
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From john.nash778 at gmail.com  Sat Jun 11 20:07:20 2016
From: john.nash778 at gmail.com (John Nash)
Date: Sat, 11 Jun 2016 21:37:20 +0530
Subject: [Freeswitch-users] Freeswitch changes payload type of codec
In-Reply-To: <14ed01d1c312$63834530$2a89cf90$@freeswitch.org>
References: <CAA4rGPy7nBRKux0ZkrBTiXz8A72cg0U1zJ+Zct+ppjXaMSpDAw@mail.gmail.com>
	<14ed01d1c312$63834530$2a89cf90$@freeswitch.org>
Message-ID: <CAA4rGPy1Kp+9G7X=em6arHOCZ=M8i9JZW0c=tJhWySG-5dv=Ow@mail.gmail.com>

Ok will fill I was not sure it was a bug.

On Fri, Jun 10, 2016 at 5:49 PM, Ken Rice <krice at freeswitch.org> wrote:

> Buts should be filed at https://freeswitch.org/jira
>
>
>
> Please Attach a complete log of a call doing this (see the note at the top
> of jira right under the menu bar on file attachments)
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *John Nash
> *Sent:* Friday, June 10, 2016 1:33 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] Freeswitch changes payload type of codec
>
>
>
> I am facing same issue described in this old post
> http://lists.freeswitch.org/pipermail/freeswitch-users/2009-June/042823.html
>
>
>
> I am using media_bypass mode.
>
>
>
> If any more information like traces etc needed I can provide.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From olegstolyar at gmail.com  Sat Jun 11 21:00:15 2016
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Sat, 11 Jun 2016 10:00:15 -0700
Subject: [Freeswitch-users] Dynamically change conference moh-sound
Message-ID: <CAC8JMOJL2jodSym_TFe23xr93xNBqkHTTLNtH0VC-v8wCmk2RA@mail.gmail.com>

Hi guys,

I need to play different hold music (local stream) to conferences depending
on where the first conference participant is calling from.  I have a couple
dozen various moh streams.

I'd rather not have to create a couple dozen almost identical conference
profiles.

Is there a way to dynamically set moh-sound on a conference in the channel
before creating the conference?
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From deepikay at iiitd.ac.in  Sat Jun 11 22:20:56 2016
From: deepikay at iiitd.ac.in (Deepika Yadav)
Date: Sat, 11 Jun 2016 23:50:56 +0530
Subject: [Freeswitch-users] Reject Inbound Call and Callback
Message-ID: <CAM99_Y=u+7gSogm1Lr-UaiAsSjYwVW0Av4xbzgh=MJuvoqdiYA@mail.gmail.com>

Hi,

I want to reject an incoming call from a mobile number in the public
context and callback to it

how should the dialplan look like:


<include>
  <extension name="public_did">
    <condition field="destination_number" expression="^(88888888)$">
<action application="set" data="domain_name=$${domain}"/>
<action application="log" data="INFO domain is $${domain}"/>
<action application="set" data="callerID = $${caller_id_number}"/>
<action application="log" data="INFO src number is $${callerID}"/>
<action application="respond" data="503"/>
<action application="bridge"
data="{sip_auth_username=1003,sip_auth_password=R at D
~o$121$@IIITD}sofia/external/1003 at 52.38.67.78 9769658019"/>
<!--<action application="respond" data="reponse_info" data="407"/> -->
<!--<action application="transfer" data="5000 XML default"/> -->

    </condition>
  </extension>
</include>






Regards,
Deepika
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From deepikay at iiitd.ac.in  Sat Jun 11 22:26:42 2016
From: deepikay at iiitd.ac.in (Deepika Yadav)
Date: Sat, 11 Jun 2016 23:56:42 +0530
Subject: [Freeswitch-users] Reject Inbound Call and Callback
In-Reply-To: <CAM99_Y=u+7gSogm1Lr-UaiAsSjYwVW0Av4xbzgh=MJuvoqdiYA@mail.gmail.com>
References: <CAM99_Y=u+7gSogm1Lr-UaiAsSjYwVW0Av4xbzgh=MJuvoqdiYA@mail.gmail.com>
Message-ID: <CAM99_Y=czCKWSmgFqfZZJ+68FkqA_z+X65BF2K18CHbgF2XGtw@mail.gmail.com>

sorry for the incomplete message, please ignore the previous message.

I am creating the dialplan in this way:

<include>
  <extension name="public_did">
    <condition field="destination_number" expression="^(88888888)$">
<action application="set" data="callerID = $${caller_id_number}"/>
<action application="log" data="INFO src number is $${callerID}"/>
        <action application="respond" data="503"/>
<action application="originate"
data="{sip_auth_username=1003,sip_auth_password=*******}sofia/external/1003@
<1003 at 52.38.67.78>X.X.X.X $callerID "/>
    </condition>
  </extension>
</include>


<extension name="show_prepare">
<condition field="destination_number" expression="^(?:\+91|0|)(\d{10})$">

<action application="bridge" data="sofia/gateway/MySIP/91$1"/>
      </condition>
</extension>

After rejecting the call, new channel for outgoing call does not start ,
also variable "callerID" is set to empty not capturing the source number

Regards,
Deepika



On Sat, Jun 11, 2016 at 11:50 PM, Deepika Yadav <deepikay at iiitd.ac.in>
wrote:

> Hi,
>
> I want to reject an incoming call from a mobile number in the public
> context and callback to it
>
> how should the dialplan look like:
>
>
> <include>
>   <extension name="public_did">
>     <condition field="destination_number" expression="^(88888888)$">
> <action application="set" data="domain_name=$${domain}"/>
> <action application="log" data="INFO domain is $${domain}"/>
> <action application="set" data="callerID = $${caller_id_number}"/>
> <action application="log" data="INFO src number is $${callerID}"/>
> <action application="respond" data="503"/>
> <action application="bridge"
> data="{sip_auth_username=1003,sip_auth_password=R at D
> ~o$121$@IIITD}sofia/external/1003 at 52.38.67.78 9769658019"/>
> <!--<action application="respond" data="reponse_info" data="407"/> -->
> <!--<action application="transfer" data="5000 XML default"/> -->
>
>     </condition>
>   </extension>
> </include>
>
>
>
>
>
>
> Regards,
> Deepika
>
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From krice at freeswitch.org  Sat Jun 11 22:42:32 2016
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 11 Jun 2016 13:42:32 -0500
Subject: [Freeswitch-users] Reject Inbound Call and Callback
In-Reply-To: <CAM99_Y=czCKWSmgFqfZZJ+68FkqA_z+X65BF2K18CHbgF2XGtw@mail.gmail.com>
References: <CAM99_Y=u+7gSogm1Lr-UaiAsSjYwVW0Av4xbzgh=MJuvoqdiYA@mail.gmail.com>
	<CAM99_Y=czCKWSmgFqfZZJ+68FkqA_z+X65BF2K18CHbgF2XGtw@mail.gmail.com>
Message-ID: <5D156F9A-BEC7-433E-BF74-A6654C417DBD@freeswitch.org>

Look at your logs you'll notice that originate is not a dialplan command and after you hang up on the incoming call the channel is dead. The correct way to do this is with a script (probably ESl) that sees the incoming call hangs it up and triggers the originate command

Sent from my iPhone

> On Jun 11, 2016, at 1:26 PM, Deepika Yadav <deepikay at iiitd.ac.in> wrote:
> 
> sorry for the incomplete message, please ignore the previous message.
> 
> I am creating the dialplan in this way:
> 
> <include>
>   <extension name="public_did">
>     <condition field="destination_number" expression="^(88888888)$">  
> 	<action application="set" data="callerID = $${caller_id_number}"/>	
> 	<action application="log" data="INFO src number is $${callerID}"/>
>         <action application="respond" data="503"/>
> 	<action application="originate" data="{sip_auth_username=1003,sip_auth_password=*******}sofia/external/1003 at X.X.X.X $callerID "/>
> 	
>     </condition>
>   </extension>
> </include>
> 
> 
> <extension name="show_prepare">
> 	<condition field="destination_number" expression="^(?:\+91|0|)(\d{10})$">
>         
> 	<action application="bridge" data="sofia/gateway/MySIP/91$1"/>
>       </condition>
> </extension>  
> 
> After rejecting the call, new channel for outgoing call does not start , also variable "callerID" is set to empty not capturing the source number
> 
> Regards,
> Deepika
> 
> 
> 
>> On Sat, Jun 11, 2016 at 11:50 PM, Deepika Yadav <deepikay at iiitd.ac.in> wrote:
>> Hi,
>> 
>> I want to reject an incoming call from a mobile number in the public context and callback to it
>> 
>> how should the dialplan look like:
>> 
>> 
>> <include>
>>   <extension name="public_did">
>>     <condition field="destination_number" expression="^(88888888)$">  
>> 	<action application="set" data="domain_name=$${domain}"/>
>> 	<action application="log" data="INFO domain is $${domain}"/>
>> 	<action application="set" data="callerID = $${caller_id_number}"/>	
>> 	<action application="log" data="INFO src number is $${callerID}"/>
>> <action application="respond" data="503"/>
>> 	<action application="bridge" data="{sip_auth_username=1003,sip_auth_password=R at D~o$121$@IIITD}sofia/external/1003 at 52.38.67.78 9769658019"/>
>> 	<!--<action application="respond" data="reponse_info" data="407"/> -->
>> 	
>> 	<!--<action application="transfer" data="5000 XML default"/> -->
>> 	
>> 	
>> 	  
>>     </condition>
>>   </extension>
>> </include>
>> 
>> 
>> 
>> 
>> 
>> 
>> Regards,
>> Deepika
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From vittico at gmail.com  Sun Jun 12 00:22:08 2016
From: vittico at gmail.com (Victor Medina)
Date: Sat, 11 Jun 2016 22:22:08 +0200
Subject: [Freeswitch-users] ICE issue
In-Reply-To: <CAL+i1s9V6ZYJYb64VNdwjgB0z83He1a7Rk8uaqnESL3_SVuiUg@mail.gmail.com>
References: <CAL+i1s-_Sg8iN+ygKS0yVhM=_cdvvtC29at3L=5hHzcWyE3f8Q@mail.gmail.com>
	<CAL+i1s8at0j19ZjUnYVve_NDLaHLh5Gzbze=KYaw5v4j7_VYTA@mail.gmail.com>
	<CAEA-bfHwFWopNNvGpYj-8oOLe9EBJ-UA4LQDX73w5fkZYx-6Gg@mail.gmail.com>
	<CAEA-bfGHCH51VZbODgTeEX5kGd1n+ZRociW9d76gwGck1UXqHw@mail.gmail.com>
	<CAL+i1s9V6ZYJYb64VNdwjgB0z83He1a7Rk8uaqnESL3_SVuiUg@mail.gmail.com>
Message-ID: <CAEA-bfG3qoeFjUzD0uFn4BfS=-JE_Z6iSJFAPHoY_9xBkkuZjQ@mail.gmail.com>

Are they enabled con FS side??
El 11/6/2016 5:39, "Bastien LE REST" <ba.lerest at gmail.com> escribi?:

> Hello Victor,
>
> Thank you for your reply. I tried to add both iLBC and Speex as a priority
> codec in the codecs list but unfortunately that didn't fix the problem.
>
> Is it normal that FS doesn't find a candidate for RTCP?
>
> Kind regards,
> Bastien.
>
> On 10 June 2016 at 19:33, Victor Medina <vittico at gmail.com> wrote:
>
>> More...
>>
>> On iOS you seem to be sending:
>>
>>
>>    1.    a=rtpmap:98 speex/16000
>>    2.    a=rtpmap:97 speex/8000
>>    3.    a=rtpmap:99 speex/32000
>>    4.    a=rtpmap:104 iLBC/8000
>>    5.    a=fmtp:104 mode=30
>>    6.    a=rtpmap:3 GSM/8000
>>    7.    a=rtpmap:0 PCMU/8000
>>    8.    a=rtpmap:8 PCMA/8000
>>    9.    a=rtpmap:9 G722/8000
>>
>>
>> While on Android I can only see:
>>
>>
>>    1.    a=rtpmap:98 speex/16000
>>    2.    a=rtpmap:97 speex/8000
>>    3.    a=rtpmap:99 speex/32000
>>    4.    a=rtpmap:104 iLBC/8000
>>
>>
>> FS is correctly complaining about it:
>>
>> 2016-06-10 12:48:33.533730 [ERR] mod_sofia.c:2346 CODEC NEGOTIATION
>> ERROR.  SDP:
>>
>>
>> Sin mas a que hacer referencia,
>>
>> Victor Medina
>>
>> On Fri, Jun 10, 2016 at 2:26 PM, Victor Medina <vittico at gmail.com> wrote:
>>
>>> There is seems to be a codec issue.
>>> Are you sure codecs are right on the Android side?
>>>
>>>
>>> Sin mas a que hacer referencia,
>>>
>>> Victor Medina
>>>
>>> On Fri, Jun 10, 2016 at 1:28 PM, Bastien LE REST <ba.lerest at gmail.com>
>>> wrote:
>>>
>>>> Bonjour,
>>>>
>>>> I thought the issue was solved since the calls work just fine now with
>>>> PJSIP on iOS, however the calls don't work with PJSIP on Android and I
>>>> don't understand why, the PJSIP configuration is exactly the same on both
>>>> devices.
>>>>
>>>> FS finds a candidate for RTP but it doesn't find a candidate for RTCP.
>>>> Please note that I only have 3 candidate.
>>>>
>>>> FS trace: https://pastebin.freeswitch.org/view/8d4df983
>>>>
>>>> Could you please have a look at the trace, and let me know what am I
>>>> doing wrong?
>>>>
>>>> Thank you very much in advance for your help in this matter.
>>>>
>>>> Kind regards,
>>>>
>>>> Bastien.
>>>>
>>>> On 4 June 2016 at 13:44, Bastien LE REST <ba.lerest at gmail.com> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> When I enable ICE using PJSIP, FreeSwitch answers "488 Not Acceptable
>>>>> Here". When I disable ICE, I don't have any problems, the calls work just
>>>>> fine. I tried to use another SIP client with ICE and the call works fine.
>>>>> So I think the problem has to do with PJSIP.
>>>>>
>>>>> I'm working on the migration from an Asterisk architecture to a
>>>>> FreeSwitch architecture. I have to deal with PJSIP and ICE support because
>>>>> my client is a mobile application who is already published on the stores.
>>>>>
>>>>> PJSIP trace : https://pastebin.freeswitch.org/view/ddeaae4b
>>>>> FreeSwitch trace: https://pastebin.freeswitch.org/view/d380e01d
>>>>>
>>>>> Could you please have a look at the trace and tell me what am I doing
>>>>> wrong? Thank you in advance.
>>>>>
>>>>> Kind regards,
>>>>> Bastien.
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From piotrek.gregor at gmail.com  Sun Jun 12 00:48:35 2016
From: piotrek.gregor at gmail.com (Piotr Gregor)
Date: Sat, 11 Jun 2016 21:48:35 +0100
Subject: [Freeswitch-users] mod_gsmopen: are USB dongles the only option?
In-Reply-To: <CACOYK=jfDnc9S63PMEje_CqZBYLJLsOi+szCuuYwANcXhLv2Mw@mail.gmail.com>
References: <CACOYK=g=DrJipsvG6jg8yqUjC+bd5rhsUfgRH7eOf+tsF14=fQ@mail.gmail.com>
	<CALXCt0qpUzz115iNjBLcTUpWCXZkpooVhL574eTC7zHM+VrDWA@mail.gmail.com>
	<CALXCt0rqW3or4=k5GSjqU4mbt2=D4SRT9Uh4m-80UWV2TOLcaA@mail.gmail.com>
	<CALXCt0qVKarWoyS5XGQf5MbRDcN-9Lq0KunJGuk2_B8gcLYaKQ@mail.gmail.com>
	<CACOYK=i03TXCcrLyLDtoy7g7gjtjSzWCCrX42GCbTTPaJsD8Bg@mail.gmail.com>
	<CAFiqYuk5zTtn7=EANngMiBzxT=eXrmwwPn=d_PRn7x2MKmTUYw@mail.gmail.com>
	<CACOYK=jfDnc9S63PMEje_CqZBYLJLsOi+szCuuYwANcXhLv2Mw@mail.gmail.com>
Message-ID: <CAO3obxb8wddg0SM3UFVbV78pwKwmXetzH_mrjFC1-eOaJPZPSA@mail.gmail.com>

Huawei MU609 with miniPCIe:

https://kamami.pl/moduly-gsmgprs/559505-huawei-mu609-modul-hspaumtsgsm-ze-zlaczem-minipcie.html

cheers,
Piotr
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From kevin.long at haloprivacy.com  Sun Jun 12 03:27:40 2016
From: kevin.long at haloprivacy.com (Kevin Long)
Date: Sat, 11 Jun 2016 23:27:40 +0000
Subject: [Freeswitch-users] Authentication of SIP phones against LDAP
	directory?
Message-ID: <A6F2618F-2966-4C32-B49A-A35CC03A8164@haloprivacy.com>



Hello,


Can Freeswitch authenticate SIP phone logins from an LDAP directory ?

Hoping to integrate this so my SIP users can use the same password/username they use for all other applications.


Regards,

Kevin Long



From miguelaustro at gmail.com  Sun Jun 12 11:47:45 2016
From: miguelaustro at gmail.com (Miguel Oyarzo)
Date: Sun, 12 Jun 2016 17:47:45 +1000
Subject: [Freeswitch-users] FS as a SRS
In-Reply-To: <CAFGRPVoGxQvn=PSK1Miq5QkpjrZWROoGmXSzf+XMh6titnct3A@mail.gmail.com>
References: <CAFGRPVqfkVxgfSrUWQXmNWEcSm5X+7_68u=x4qt-pgNXo-PbLw@mail.gmail.com>
	<CA+d2mV-5pG-XCy1w=sCNk5gMduy6row1YYM+7zS7dkfomqyq2w@mail.gmail.com>
	<CAFGRPVoGxQvn=PSK1Miq5QkpjrZWROoGmXSzf+XMh6titnct3A@mail.gmail.com>
Message-ID: <CA+d2mV-P13CYNOUedJ9qKOU1RU4heL13Pzg8=agYyZk5v0phDg@mail.gmail.com>

Hi David,
I got you, but FS can't accept multiple m=audio lines, which makes it
incompatible for SIPREC.
I don't think there is plans to modify the SIP diver in freeswitch, in
order to
support this protocol. Enabling additional RTP engines for a single SDP
session is not trivial.

Regards,



On Wed, Jun 8, 2016 at 10:03 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello Miguel,
>
> Thanks for your reply, but I think maybe I didn't explain myself
> correctly, let me try again:
>
> A recording server receives INVITEs including multiple "m=" which will be
> used as rtp source so the server can record an ongoing conversation
> happening somewhere else. As per RFC4566 an SDP may have as many m= as it
> wants (https://tools.ietf.org/html/rfc4566#page-22).
>
> The idea is to use SIPREC the following way:
>
> The voice server will choose the call to record and send a 3rd leg to the
> recording server, which will receive the INVITE ans answer with 200 OK and
> start recording the rtps.
>
> Regards,
>
> David
>
> On Wed, Jun 8, 2016 at 12:40 PM, Miguel Oyarzo <miguelaustro at gmail.com>
> wrote:
>
>>
>>
>> Nighter FS nor Asterisk will process two same-type RTP streams in the
>> same SDP. The 2nd RTP streams will be declined and answered with a port 0
>> (200OK), as suggested by rfc3264.
>>
>> Regards,
>>
>> On Wed, Jun 8, 2016 at 2:37 AM, David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> Hello,
>>> Is it possible to use FS as a recording server?
>>>
>>> We will be receiving INVITEs with a siprec tag and sdp with the
>>> participants. To which FS should answer with 2 rtp ports on which to
>>> receive the audio. We would be receiving 2 rtp streams that we need to
>>> record to a file.
>>>
>>> Thanks!
>>>
>>> David
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> "Do not use a cannon to kill a mosquito" (Confucius)
>>
>> Miguel Oyarzo
>> VoIP Engineer && DevOps
>> Linux User: # 483188 - counter.li.org
>> http://au.linkedin.com/in/mikeaustralia
>> Melbourne, Australia
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
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> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
"Do not use a cannon to kill a mosquito" (Confucius)

Miguel Oyarzo
VoIP Engineer && DevOps
Linux User: # 483188 - counter.li.org
http://au.linkedin.com/in/mikeaustralia
Melbourne, Australia
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From ba.lerest at gmail.com  Sun Jun 12 14:22:29 2016
From: ba.lerest at gmail.com (Bastien LE REST)
Date: Sun, 12 Jun 2016 11:22:29 +0100
Subject: [Freeswitch-users] ICE issue
In-Reply-To: <CAEA-bfG3qoeFjUzD0uFn4BfS=-JE_Z6iSJFAPHoY_9xBkkuZjQ@mail.gmail.com>
References: <CAL+i1s-_Sg8iN+ygKS0yVhM=_cdvvtC29at3L=5hHzcWyE3f8Q@mail.gmail.com>
	<CAL+i1s8at0j19ZjUnYVve_NDLaHLh5Gzbze=KYaw5v4j7_VYTA@mail.gmail.com>
	<CAEA-bfHwFWopNNvGpYj-8oOLe9EBJ-UA4LQDX73w5fkZYx-6Gg@mail.gmail.com>
	<CAEA-bfGHCH51VZbODgTeEX5kGd1n+ZRociW9d76gwGck1UXqHw@mail.gmail.com>
	<CAL+i1s9V6ZYJYb64VNdwjgB0z83He1a7Rk8uaqnESL3_SVuiUg@mail.gmail.com>
	<CAEA-bfG3qoeFjUzD0uFn4BfS=-JE_Z6iSJFAPHoY_9xBkkuZjQ@mail.gmail.com>
Message-ID: <CAL+i1s9RFUry9XEzGAp81jJcPp-VdfNX=BtZ-=8UufdyhRum2Q@mail.gmail.com>

Hello Victor,

Yes, I configured FS to only use iLBC codec with mod_iLBC. iOS works fine
using this codec but I still have the same problem with android.

Here's the trace: https://pastebin.freeswitch.org/view/fcf3b334

Thank you for your help, it's much appreciated.

Kind regards,
Bastien.

On 11 June 2016 at 21:22, Victor Medina <vittico at gmail.com> wrote:

> Are they enabled con FS side??
> El 11/6/2016 5:39, "Bastien LE REST" <ba.lerest at gmail.com> escribi?:
>
>> Hello Victor,
>>
>> Thank you for your reply. I tried to add both iLBC and Speex as a
>> priority codec in the codecs list but unfortunately that didn't fix the
>> problem.
>>
>> Is it normal that FS doesn't find a candidate for RTCP?
>>
>> Kind regards,
>> Bastien.
>>
>> On 10 June 2016 at 19:33, Victor Medina <vittico at gmail.com> wrote:
>>
>>> More...
>>>
>>> On iOS you seem to be sending:
>>>
>>>
>>>    1.    a=rtpmap:98 speex/16000
>>>    2.    a=rtpmap:97 speex/8000
>>>    3.    a=rtpmap:99 speex/32000
>>>    4.    a=rtpmap:104 iLBC/8000
>>>    5.    a=fmtp:104 mode=30
>>>    6.    a=rtpmap:3 GSM/8000
>>>    7.    a=rtpmap:0 PCMU/8000
>>>    8.    a=rtpmap:8 PCMA/8000
>>>    9.    a=rtpmap:9 G722/8000
>>>
>>>
>>> While on Android I can only see:
>>>
>>>
>>>    1.    a=rtpmap:98 speex/16000
>>>    2.    a=rtpmap:97 speex/8000
>>>    3.    a=rtpmap:99 speex/32000
>>>    4.    a=rtpmap:104 iLBC/8000
>>>
>>>
>>> FS is correctly complaining about it:
>>>
>>> 2016-06-10 12:48:33.533730 [ERR] mod_sofia.c:2346 CODEC NEGOTIATION
>>> ERROR.  SDP:
>>>
>>>
>>> Sin mas a que hacer referencia,
>>>
>>> Victor Medina
>>>
>>> On Fri, Jun 10, 2016 at 2:26 PM, Victor Medina <vittico at gmail.com>
>>> wrote:
>>>
>>>> There is seems to be a codec issue.
>>>> Are you sure codecs are right on the Android side?
>>>>
>>>>
>>>> Sin mas a que hacer referencia,
>>>>
>>>> Victor Medina
>>>>
>>>> On Fri, Jun 10, 2016 at 1:28 PM, Bastien LE REST <ba.lerest at gmail.com>
>>>> wrote:
>>>>
>>>>> Bonjour,
>>>>>
>>>>> I thought the issue was solved since the calls work just fine now with
>>>>> PJSIP on iOS, however the calls don't work with PJSIP on Android and I
>>>>> don't understand why, the PJSIP configuration is exactly the same on both
>>>>> devices.
>>>>>
>>>>> FS finds a candidate for RTP but it doesn't find a candidate for RTCP.
>>>>> Please note that I only have 3 candidate.
>>>>>
>>>>> FS trace: https://pastebin.freeswitch.org/view/8d4df983
>>>>>
>>>>> Could you please have a look at the trace, and let me know what am I
>>>>> doing wrong?
>>>>>
>>>>> Thank you very much in advance for your help in this matter.
>>>>>
>>>>> Kind regards,
>>>>>
>>>>> Bastien.
>>>>>
>>>>> On 4 June 2016 at 13:44, Bastien LE REST <ba.lerest at gmail.com> wrote:
>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> When I enable ICE using PJSIP, FreeSwitch answers "488 Not Acceptable
>>>>>> Here". When I disable ICE, I don't have any problems, the calls work just
>>>>>> fine. I tried to use another SIP client with ICE and the call works fine.
>>>>>> So I think the problem has to do with PJSIP.
>>>>>>
>>>>>> I'm working on the migration from an Asterisk architecture to a
>>>>>> FreeSwitch architecture. I have to deal with PJSIP and ICE support because
>>>>>> my client is a mobile application who is already published on the stores.
>>>>>>
>>>>>> PJSIP trace : https://pastebin.freeswitch.org/view/ddeaae4b
>>>>>> FreeSwitch trace: https://pastebin.freeswitch.org/view/d380e01d
>>>>>>
>>>>>> Could you please have a look at the trace and tell me what am I doing
>>>>>> wrong? Thank you in advance.
>>>>>>
>>>>>> Kind regards,
>>>>>> Bastien.
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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From fanx07 at gmail.com  Sun Jun 12 18:17:43 2016
From: fanx07 at gmail.com (Anonim Stefan)
Date: Sun, 12 Jun 2016 17:17:43 +0300
Subject: [Freeswitch-users] Dynamic Group Call
In-Reply-To: <12523881-DD17-4269-A9D9-CB7CF28B18C1@jerris.com>
References: <CAL82Oa1YksiVpFFxovF8ghy_3XE_Rkp45wisj9Zt4Mnp0jMnnA@mail.gmail.com>
	<12523881-DD17-4269-A9D9-CB7CF28B18C1@jerris.com>
Message-ID: <CAO7wsJvYwf701R2=SAB67QGqS+E=B6h=+CgNWatwwqsesC9hZw@mail.gmail.com>

Thank you for the answer Michael.

As an alternative to lua, I have managed to do that using mod_python, with
<application="python" data="my_python_script_that
_adds_conference_auto_call">.

Stefan

On Fri, Jun 10, 2016 at 8:19 PM, Michael Jerris <mike at jerris.com> wrote:

> not easy to do in xml dialplan but probably doable in any of the dynamic
> dialplan bindings via xml_curl or one of the embedded langs like lua
>
> On Jun 10, 2016, at 6:05 AM, Mititelu Stefan <stefan.mititelu92 at gmail.com>
> wrote:
>
> Hi,
>
> I'm a Freeswitch newbee and I'm trying to set up dynamic group calls.
>
> Freeswitch will receive a header with "Group-Users:user1,user2.." and what
> I want to do is to "transfer" to a dinamycally created context that will
> look like:
>       <action application="conference_set_auto_outcall" data="user1"/>
>       <action application="conference_set_auto_outcall" data="user2"/>
>
> Is this possible? Can you give me some references?
>
> From what I see in some examples one can transfer only to existent XML
> context:
>     <action application="transfer" data="1105 XML default"/>
>
> Thanks,
> Stefan
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ssinyagin at gmail.com  Sun Jun 12 18:18:49 2016
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sun, 12 Jun 2016 16:18:49 +0200
Subject: [Freeswitch-users] Authentication of SIP phones against LDAP
	directory?
In-Reply-To: <A6F2618F-2966-4C32-B49A-A35CC03A8164@haloprivacy.com>
References: <A6F2618F-2966-4C32-B49A-A35CC03A8164@haloprivacy.com>
Message-ID: <CACOYK=jK6w8xzuTdgpPAto-Zd-W71H6np0R2xKFfyj1Qxoz=7g@mail.gmail.com>

there's a principal difference in how authentication works in SIP and LDAP:

SIP is using challenge-response, so the server needs either the
cleartext password, or A1-hash.

LDAP is primarily using salted hashes to store the passwords, so
there's no way to retrieve the cleartext password.

So, the best you can do, is to have a separate LDAP attribute for the
SIP password, and keep clear text passwords in it. But then it comes
to the same problem as before, that the users have to maintain two
different passwords.




On Sun, Jun 12, 2016 at 1:27 AM, Kevin Long <kevin.long at haloprivacy.com> wrote:
>
>
> Hello,
>
>
> Can Freeswitch authenticate SIP phone logins from an LDAP directory ?
>
> Hoping to integrate this so my SIP users can use the same password/username they use for all other applications.
>
>
> Regards,
>
> Kevin Long
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From ssinyagin at gmail.com  Sun Jun 12 18:21:06 2016
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sun, 12 Jun 2016 16:21:06 +0200
Subject: [Freeswitch-users] mod_gsmopen: are USB dongles the only option?
In-Reply-To: <CAO3obxb8wddg0SM3UFVbV78pwKwmXetzH_mrjFC1-eOaJPZPSA@mail.gmail.com>
References: <CACOYK=g=DrJipsvG6jg8yqUjC+bd5rhsUfgRH7eOf+tsF14=fQ@mail.gmail.com>
	<CALXCt0qpUzz115iNjBLcTUpWCXZkpooVhL574eTC7zHM+VrDWA@mail.gmail.com>
	<CALXCt0rqW3or4=k5GSjqU4mbt2=D4SRT9Uh4m-80UWV2TOLcaA@mail.gmail.com>
	<CALXCt0qVKarWoyS5XGQf5MbRDcN-9Lq0KunJGuk2_B8gcLYaKQ@mail.gmail.com>
	<CACOYK=i03TXCcrLyLDtoy7g7gjtjSzWCCrX42GCbTTPaJsD8Bg@mail.gmail.com>
	<CAFiqYuk5zTtn7=EANngMiBzxT=eXrmwwPn=d_PRn7x2MKmTUYw@mail.gmail.com>
	<CACOYK=jfDnc9S63PMEje_CqZBYLJLsOi+szCuuYwANcXhLv2Mw@mail.gmail.com>
	<CAO3obxb8wddg0SM3UFVbV78pwKwmXetzH_mrjFC1-eOaJPZPSA@mail.gmail.com>
Message-ID: <CACOYK=hyjjmgNSnoeQiWKH-ZtvdAStH1iffSCC1Ogpe4n2c0Nw@mail.gmail.com>

yes, I know it exists. Also there's a detailed documentation available
at Huawei website, and it tells the pins where PCM audio is
transported. Those pins are "reserved" in mini-PCIE standard, so your
PCB will not have them wired.



On Sat, Jun 11, 2016 at 10:48 PM, Piotr Gregor <piotrek.gregor at gmail.com> wrote:
> Huawei MU609 with miniPCIe:
>
> https://kamami.pl/moduly-gsmgprs/559505-huawei-mu609-modul-hspaumtsgsm-ze-zlaczem-minipcie.html
>
> cheers,
> Piotr
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From ashwinrath at gmail.com  Sun Jun 12 18:26:58 2016
From: ashwinrath at gmail.com (Ashwin Rath)
Date: Sun, 12 Jun 2016 19:56:58 +0530
Subject: [Freeswitch-users] Auto record by call direction
Message-ID: <CAA_Nwu7hLPpK4p0CyAbpCfPcLK4BRAg5Mz5xu4YKgAK4g=Vwbw@mail.gmail.com>

Hello


To record a call (inbound or outbound) i set the following variable for a
user in the directory folder

 <variable name="user_record" value="enabled"/>

However i would like to record inbound calls only. Is there some way to
selectively auto record inbound calls only ?
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From deepikay at iiitd.ac.in  Sun Jun 12 18:33:28 2016
From: deepikay at iiitd.ac.in (Deepika Yadav)
Date: Sun, 12 Jun 2016 20:03:28 +0530
Subject: [Freeswitch-users] Reject Inbound Call and Callback
In-Reply-To: <5D156F9A-BEC7-433E-BF74-A6654C417DBD@freeswitch.org>
References: <CAM99_Y=u+7gSogm1Lr-UaiAsSjYwVW0Av4xbzgh=MJuvoqdiYA@mail.gmail.com>
	<CAM99_Y=czCKWSmgFqfZZJ+68FkqA_z+X65BF2K18CHbgF2XGtw@mail.gmail.com>
	<5D156F9A-BEC7-433E-BF74-A6654C417DBD@freeswitch.org>
Message-ID: <CAM99_Y=SVtSeQZVXGoCZ9fZMDfVthH3JJtVTTbN3OXxpgcQBOA@mail.gmail.com>

Even in ESL, first I have to reject the call and start a new session.

What is the command to reject a call in mod_python, I tried
session.respond() that did not work.


Python script :

def handler(session, args):

    namer = session.getVariable('caller_id_number')
    freeswitch.consoleLog('info','Called by '+namer)
    session.respond()
    s = freeswitch.Session();
    s.originate(session,"
{sip_auth_username=1002,sip_auth_password=XXXXXXX}sofia/external/1002 at XX.XX.XX.XX
 9868789909")













On Sun, Jun 12, 2016 at 12:12 AM, Ken Rice <krice at freeswitch.org> wrote:

> Look at your logs you'll notice that originate is not a dialplan command
> and after you hang up on the incoming call the channel is dead. The correct
> way to do this is with a script (probably ESl) that sees the incoming call
> hangs it up and triggers the originate command
>
> Sent from my iPhone
>
> On Jun 11, 2016, at 1:26 PM, Deepika Yadav <deepikay at iiitd.ac.in> wrote:
>
> sorry for the incomplete message, please ignore the previous message.
>
> I am creating the dialplan in this way:
>
> <include>
>   <extension name="public_did">
>     <condition field="destination_number" expression="^(88888888)$">
> <action application="set" data="callerID = $${caller_id_number}"/>
> <action application="log" data="INFO src number is $${callerID}"/>
>         <action application="respond" data="503"/>
> <action application="originate"
> data="{sip_auth_username=1003,sip_auth_password=*******}sofia/external/
> 1003@ <1003 at 52.38.67.78>X.X.X.X $callerID "/>
>     </condition>
>   </extension>
> </include>
>
>
> <extension name="show_prepare">
> <condition field="destination_number" expression="^(?:\+91|0|)(\d{10})$">
>
> <action application="bridge" data="sofia/gateway/MySIP/91$1"/>
>       </condition>
> </extension>
>
> After rejecting the call, new channel for outgoing call does not start ,
> also variable "callerID" is set to empty not capturing the source number
>
> Regards,
> Deepika
>
>
>
> On Sat, Jun 11, 2016 at 11:50 PM, Deepika Yadav <deepikay at iiitd.ac.in>
> wrote:
>
>> Hi,
>>
>> I want to reject an incoming call from a mobile number in the public
>> context and callback to it
>>
>> how should the dialplan look like:
>>
>>
>> <include>
>>   <extension name="public_did">
>>     <condition field="destination_number" expression="^(88888888)$">
>> <action application="set" data="domain_name=$${domain}"/>
>> <action application="log" data="INFO domain is $${domain}"/>
>> <action application="set" data="callerID = $${caller_id_number}"/>
>> <action application="log" data="INFO src number is $${callerID}"/>
>> <action application="respond" data="503"/>
>> <action application="bridge"
>> data="{sip_auth_username=1003,sip_auth_password=R at D
>> ~o$121$@IIITD}sofia/external/1003 at 52.38.67.78 9769658019"/>
>> <!--<action application="respond" data="reponse_info" data="407"/> -->
>> <!--<action application="transfer" data="5000 XML default"/> -->
>>
>>     </condition>
>>   </extension>
>> </include>
>>
>>
>>
>>
>>
>>
>> Regards,
>> Deepika
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ashwinrath at gmail.com  Sun Jun 12 18:53:15 2016
From: ashwinrath at gmail.com (Ashwin Rath)
Date: Sun, 12 Jun 2016 20:23:15 +0530
Subject: [Freeswitch-users] Auto record by call direction
In-Reply-To: <CAA_Nwu7hLPpK4p0CyAbpCfPcLK4BRAg5Mz5xu4YKgAK4g=Vwbw@mail.gmail.com>
References: <CAA_Nwu7hLPpK4p0CyAbpCfPcLK4BRAg5Mz5xu4YKgAK4g=Vwbw@mail.gmail.com>
Message-ID: <CAA_Nwu4Y9vru8MEPE5dkzwaEanjQ3DpndvF9z8qT4S0=vCwzPw@mail.gmail.com>

Just found out the latest FusionPBX has this variable extended to specify
call direction.

On Sun, Jun 12, 2016 at 7:56 PM, Ashwin Rath <ashwinrath at gmail.com> wrote:

>
> Hello
>
>
> To record a call (inbound or outbound) i set the following variable for a
> user in the directory folder
>
>  <variable name="user_record" value="enabled"/>
>
> However i would like to record inbound calls only. Is there some way to
> selectively auto record inbound calls only ?
>



-- 
Ashwin Kumar Rath
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From aqsyounas at gmail.com  Sun Jun 12 19:30:52 2016
From: aqsyounas at gmail.com (Aqs Younas)
Date: Sun, 12 Jun 2016 20:30:52 +0500
Subject: [Freeswitch-users] Reject Inbound Call and Callback
In-Reply-To: <CAM99_Y=SVtSeQZVXGoCZ9fZMDfVthH3JJtVTTbN3OXxpgcQBOA@mail.gmail.com>
References: <CAM99_Y=u+7gSogm1Lr-UaiAsSjYwVW0Av4xbzgh=MJuvoqdiYA@mail.gmail.com>
	<CAM99_Y=czCKWSmgFqfZZJ+68FkqA_z+X65BF2K18CHbgF2XGtw@mail.gmail.com>
	<5D156F9A-BEC7-433E-BF74-A6654C417DBD@freeswitch.org>
	<CAM99_Y=SVtSeQZVXGoCZ9fZMDfVthH3JJtVTTbN3OXxpgcQBOA@mail.gmail.com>
Message-ID: <CAGVOjVU4VcoO=62kpTBxyopOO1z1Vq85kHENd9jL+p5fQtd7vw@mail.gmail.com>

You can hangup the call with reason.
https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_hangup
On Jun 12, 2016 7:36 PM, "Deepika Yadav" <deepikay at iiitd.ac.in> wrote:

> Even in ESL, first I have to reject the call and start a new session.
>
> What is the command to reject a call in mod_python, I tried
> session.respond() that did not work.
>
>
> Python script :
>
> def handler(session, args):
>
>     namer = session.getVariable('caller_id_number')
>     freeswitch.consoleLog('info','Called by '+namer)
>     session.respond()
>     s = freeswitch.Session();
>     s.originate(session,"
> {sip_auth_username=1002,sip_auth_password=XXXXXXX}sofia/external/1002 at XX.XX.XX.XX
>  9868789909")
>
>
>
>
>
>
>
>
>
>
>
>
>
> On Sun, Jun 12, 2016 at 12:12 AM, Ken Rice <krice at freeswitch.org> wrote:
>
>> Look at your logs you'll notice that originate is not a dialplan command
>> and after you hang up on the incoming call the channel is dead. The correct
>> way to do this is with a script (probably ESl) that sees the incoming call
>> hangs it up and triggers the originate command
>>
>> Sent from my iPhone
>>
>> On Jun 11, 2016, at 1:26 PM, Deepika Yadav <deepikay at iiitd.ac.in> wrote:
>>
>> sorry for the incomplete message, please ignore the previous message.
>>
>> I am creating the dialplan in this way:
>>
>> <include>
>>   <extension name="public_did">
>>     <condition field="destination_number" expression="^(88888888)$">
>> <action application="set" data="callerID = $${caller_id_number}"/>
>> <action application="log" data="INFO src number is $${callerID}"/>
>>         <action application="respond" data="503"/>
>> <action application="originate"
>> data="{sip_auth_username=1003,sip_auth_password=*******}sofia/external/
>> 1003@ <1003 at 52.38.67.78>X.X.X.X $callerID "/>
>>     </condition>
>>   </extension>
>> </include>
>>
>>
>> <extension name="show_prepare">
>> <condition field="destination_number" expression="^(?:\+91|0|)(\d{10})$">
>>
>> <action application="bridge" data="sofia/gateway/MySIP/91$1"/>
>>       </condition>
>> </extension>
>>
>> After rejecting the call, new channel for outgoing call does not start ,
>> also variable "callerID" is set to empty not capturing the source number
>>
>> Regards,
>> Deepika
>>
>>
>>
>> On Sat, Jun 11, 2016 at 11:50 PM, Deepika Yadav <deepikay at iiitd.ac.in>
>> wrote:
>>
>>> Hi,
>>>
>>> I want to reject an incoming call from a mobile number in the public
>>> context and callback to it
>>>
>>> how should the dialplan look like:
>>>
>>>
>>> <include>
>>>   <extension name="public_did">
>>>     <condition field="destination_number" expression="^(88888888)$">
>>> <action application="set" data="domain_name=$${domain}"/>
>>> <action application="log" data="INFO domain is $${domain}"/>
>>> <action application="set" data="callerID = $${caller_id_number}"/>
>>> <action application="log" data="INFO src number is $${callerID}"/>
>>> <action application="respond" data="503"/>
>>> <action application="bridge"
>>> data="{sip_auth_username=1003,sip_auth_password=R at D
>>> ~o$121$@IIITD}sofia/external/1003 at 52.38.67.78 9769658019"/>
>>> <!--<action application="respond" data="reponse_info" data="407"/> -->
>>> <!--<action application="transfer" data="5000 XML default"/> -->
>>>
>>>     </condition>
>>>   </extension>
>>> </include>
>>>
>>>
>>>
>>>
>>>
>>>
>>> Regards,
>>> Deepika
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From deepikay at iiitd.ac.in  Sun Jun 12 21:01:59 2016
From: deepikay at iiitd.ac.in (Deepika Yadav)
Date: Sun, 12 Jun 2016 22:31:59 +0530
Subject: [Freeswitch-users] Reject Inbound Call and Callback
In-Reply-To: <CAGVOjVU4VcoO=62kpTBxyopOO1z1Vq85kHENd9jL+p5fQtd7vw@mail.gmail.com>
References: <CAM99_Y=u+7gSogm1Lr-UaiAsSjYwVW0Av4xbzgh=MJuvoqdiYA@mail.gmail.com>
	<CAM99_Y=czCKWSmgFqfZZJ+68FkqA_z+X65BF2K18CHbgF2XGtw@mail.gmail.com>
	<5D156F9A-BEC7-433E-BF74-A6654C417DBD@freeswitch.org>
	<CAM99_Y=SVtSeQZVXGoCZ9fZMDfVthH3JJtVTTbN3OXxpgcQBOA@mail.gmail.com>
	<CAGVOjVU4VcoO=62kpTBxyopOO1z1Vq85kHENd9jL+p5fQtd7vw@mail.gmail.com>
Message-ID: <CAM99_Ymvih91h0aT06PVodK72THPXyR9upJJTM_EqyfZrak2cg@mail.gmail.com>

Even if I use Hangup command dialplan won't execute the statements after
hangup.

My basic requirement is when Freeswitch receives a call on an extension, it
should reject it and then callback.

On Sun, Jun 12, 2016 at 9:00 PM, Aqs Younas <aqsyounas at gmail.com> wrote:

> You can hangup the call with reason.
> https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_hangup
> On Jun 12, 2016 7:36 PM, "Deepika Yadav" <deepikay at iiitd.ac.in> wrote:
>
>> Even in ESL, first I have to reject the call and start a new session.
>>
>> What is the command to reject a call in mod_python, I tried
>> session.respond() that did not work.
>>
>>
>> Python script :
>>
>> def handler(session, args):
>>
>>     namer = session.getVariable('caller_id_number')
>>     freeswitch.consoleLog('info','Called by '+namer)
>>     session.respond()
>>     s = freeswitch.Session();
>>     s.originate(session,"
>> {sip_auth_username=1002,sip_auth_password=XXXXXXX}sofia/external/1002 at XX.XX.XX.XX
>>  9868789909")
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> On Sun, Jun 12, 2016 at 12:12 AM, Ken Rice <krice at freeswitch.org> wrote:
>>
>>> Look at your logs you'll notice that originate is not a dialplan command
>>> and after you hang up on the incoming call the channel is dead. The correct
>>> way to do this is with a script (probably ESl) that sees the incoming call
>>> hangs it up and triggers the originate command
>>>
>>> Sent from my iPhone
>>>
>>> On Jun 11, 2016, at 1:26 PM, Deepika Yadav <deepikay at iiitd.ac.in> wrote:
>>>
>>> sorry for the incomplete message, please ignore the previous message.
>>>
>>> I am creating the dialplan in this way:
>>>
>>> <include>
>>>   <extension name="public_did">
>>>     <condition field="destination_number" expression="^(88888888)$">
>>> <action application="set" data="callerID = $${caller_id_number}"/>
>>> <action application="log" data="INFO src number is $${callerID}"/>
>>>         <action application="respond" data="503"/>
>>> <action application="originate"
>>> data="{sip_auth_username=1003,sip_auth_password=*******}sofia/external/
>>> 1003@ <1003 at 52.38.67.78>X.X.X.X $callerID "/>
>>>     </condition>
>>>   </extension>
>>> </include>
>>>
>>>
>>> <extension name="show_prepare">
>>> <condition field="destination_number" expression="^(?:\+91|0|)(\d{10})$">
>>>
>>> <action application="bridge" data="sofia/gateway/MySIP/91$1"/>
>>>       </condition>
>>> </extension>
>>>
>>> After rejecting the call, new channel for outgoing call does not start ,
>>> also variable "callerID" is set to empty not capturing the source number
>>>
>>> Regards,
>>> Deepika
>>>
>>>
>>>
>>> On Sat, Jun 11, 2016 at 11:50 PM, Deepika Yadav <deepikay at iiitd.ac.in>
>>> wrote:
>>>
>>>> Hi,
>>>>
>>>> I want to reject an incoming call from a mobile number in the public
>>>> context and callback to it
>>>>
>>>> how should the dialplan look like:
>>>>
>>>>
>>>> <include>
>>>>   <extension name="public_did">
>>>>     <condition field="destination_number" expression="^(88888888)$">
>>>> <action application="set" data="domain_name=$${domain}"/>
>>>> <action application="log" data="INFO domain is $${domain}"/>
>>>> <action application="set" data="callerID = $${caller_id_number}"/>
>>>> <action application="log" data="INFO src number is $${callerID}"/>
>>>> <action application="respond" data="503"/>
>>>> <action application="bridge"
>>>> data="{sip_auth_username=1003,sip_auth_password=R at D
>>>> ~o$121$@IIITD}sofia/external/1003 at 52.38.67.78 9769658019"/>
>>>> <!--<action application="respond" data="reponse_info" data="407"/> -->
>>>> <!--<action application="transfer" data="5000 XML default"/> -->
>>>>
>>>>     </condition>
>>>>   </extension>
>>>> </include>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> Regards,
>>>> Deepika
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kevin.long at haloprivacy.com  Sun Jun 12 22:41:27 2016
From: kevin.long at haloprivacy.com (Kevin Long)
Date: Sun, 12 Jun 2016 18:41:27 +0000
Subject: [Freeswitch-users] Authentication of SIP phones against
	LDAP	directory?
In-Reply-To: <CACOYK=jK6w8xzuTdgpPAto-Zd-W71H6np0R2xKFfyj1Qxoz=7g@mail.gmail.com>
References: <A6F2618F-2966-4C32-B49A-A35CC03A8164@haloprivacy.com>
	<CACOYK=jK6w8xzuTdgpPAto-Zd-W71H6np0R2xKFfyj1Qxoz=7g@mail.gmail.com>
Message-ID: <F222D7AB-7728-42E7-A1BF-B1D761726FA5@haloprivacy.com>


Hi Stanislav,  thanks for the response.

Even if I do need some separate attributes, plaintext or A1 hashed,  does the functionality exist in Freeswitch to do the authentication from LDAP from these attributes?

My goal here is to run Freeswitch via Docker, because I intend to deploy it many, many times for small groups of users in an ephemeral way.  So the configuration files, including the users extensions etc,  would not be permanent and need to be generated on-the-fly when the instance is booted up.


If there is another way people are doing this kind of thing,  I would love to know.  


Thanks again,

Kevin Long


> On Jun 12, 2016, at 7:18 AM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
> 
> there's a principal difference in how authentication works in SIP and LDAP:
> 
> SIP is using challenge-response, so the server needs either the
> cleartext password, or A1-hash.
> 
> LDAP is primarily using salted hashes to store the passwords, so
> there's no way to retrieve the cleartext password.
> 
> So, the best you can do, is to have a separate LDAP attribute for the
> SIP password, and keep clear text passwords in it. But then it comes
> to the same problem as before, that the users have to maintain two
> different passwords.
> 
> 
> 
> 
> On Sun, Jun 12, 2016 at 1:27 AM, Kevin Long <kevin.long at haloprivacy.com> wrote:
>> 
>> 
>> Hello,
>> 
>> 
>> Can Freeswitch authenticate SIP phone logins from an LDAP directory ?
>> 
>> Hoping to integrate this so my SIP users can use the same password/username they use for all other applications.
>> 
>> 
>> Regards,
>> 
>> Kevin Long
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From krice at freeswitch.org  Sun Jun 12 23:02:47 2016
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 12 Jun 2016 14:02:47 -0500
Subject: [Freeswitch-users] Reject Inbound Call and Callback
In-Reply-To: <CAM99_Ymvih91h0aT06PVodK72THPXyR9upJJTM_EqyfZrak2cg@mail.gmail.com>
References: <CAM99_Y=u+7gSogm1Lr-UaiAsSjYwVW0Av4xbzgh=MJuvoqdiYA@mail.gmail.com>
	<CAM99_Y=czCKWSmgFqfZZJ+68FkqA_z+X65BF2K18CHbgF2XGtw@mail.gmail.com>
	<5D156F9A-BEC7-433E-BF74-A6654C417DBD@freeswitch.org>
	<CAM99_Y=SVtSeQZVXGoCZ9fZMDfVthH3JJtVTTbN3OXxpgcQBOA@mail.gmail.com>
	<CAGVOjVU4VcoO=62kpTBxyopOO1z1Vq85kHENd9jL+p5fQtd7vw@mail.gmail.com>
	<CAM99_Ymvih91h0aT06PVodK72THPXyR9upJJTM_EqyfZrak2cg@mail.gmail.com>
Message-ID: <C4862E63-64CD-4DA9-A30A-99FFAF7F6D06@freeswitch.org>

Ok let me try this again. Once the incoming call is hung up, the call is dead and anything processing on the channel (except the CDR) is done. No further commands in the dial-plan will execute.


If you need pro help with this contact consulting at freeswitch.org and one of us pros can knock this out pretty fast for you

Sent from my iPhone

> On Jun 12, 2016, at 12:01 PM, Deepika Yadav <deepikay at iiitd.ac.in> wrote:
> 
> Even if I use Hangup command dialplan won't execute the statements after hangup.
> 
> My basic requirement is when Freeswitch receives a call on an extension, it should reject it and then callback.
> 
>> On Sun, Jun 12, 2016 at 9:00 PM, Aqs Younas <aqsyounas at gmail.com> wrote:
>> You can hangup the call with reason.
>> https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_hangup
>> 
>>> On Jun 12, 2016 7:36 PM, "Deepika Yadav" <deepikay at iiitd.ac.in> wrote:
>>> Even in ESL, first I have to reject the call and start a new session.
>>> 
>>> What is the command to reject a call in mod_python, I tried session.respond() that did not work.
>>> 
>>> 
>>> Python script :
>>> 
>>> def handler(session, args):
>>> 
>>>     namer = session.getVariable('caller_id_number')
>>>     freeswitch.consoleLog('info','Called by '+namer)
>>>     session.respond()
>>>     s = freeswitch.Session();
>>>     s.originate(session," {sip_auth_username=1002,sip_auth_password=XXXXXXX}sofia/external/1002 at XX.XX.XX.XX  9868789909")
>>>    
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>>> On Sun, Jun 12, 2016 at 12:12 AM, Ken Rice <krice at freeswitch.org> wrote:
>>>> Look at your logs you'll notice that originate is not a dialplan command and after you hang up on the incoming call the channel is dead. The correct way to do this is with a script (probably ESl) that sees the incoming call hangs it up and triggers the originate command
>>>> 
>>>> Sent from my iPhone
>>>> 
>>>>> On Jun 11, 2016, at 1:26 PM, Deepika Yadav <deepikay at iiitd.ac.in> wrote:
>>>>> 
>>>>> sorry for the incomplete message, please ignore the previous message.
>>>>> 
>>>>> I am creating the dialplan in this way:
>>>>> 
>>>>> <include>
>>>>>   <extension name="public_did">
>>>>>     <condition field="destination_number" expression="^(88888888)$">  
>>>>> 	<action application="set" data="callerID = $${caller_id_number}"/>	
>>>>> 	<action application="log" data="INFO src number is $${callerID}"/>
>>>>>         <action application="respond" data="503"/>
>>>>> 	<action application="originate" data="{sip_auth_username=1003,sip_auth_password=*******}sofia/external/1003 at X.X.X.X $callerID "/>
>>>>> 	
>>>>>     </condition>
>>>>>   </extension>
>>>>> </include>
>>>>> 
>>>>> 
>>>>> <extension name="show_prepare">
>>>>> 	<condition field="destination_number" expression="^(?:\+91|0|)(\d{10})$">
>>>>>         
>>>>> 	<action application="bridge" data="sofia/gateway/MySIP/91$1"/>
>>>>>       </condition>
>>>>> </extension>  
>>>>> 
>>>>> After rejecting the call, new channel for outgoing call does not start , also variable "callerID" is set to empty not capturing the source number
>>>>> 
>>>>> Regards,
>>>>> Deepika
>>>>> 
>>>>> 
>>>>> 
>>>>>> On Sat, Jun 11, 2016 at 11:50 PM, Deepika Yadav <deepikay at iiitd.ac.in> wrote:
>>>>>> Hi,
>>>>>> 
>>>>>> I want to reject an incoming call from a mobile number in the public context and callback to it
>>>>>> 
>>>>>> how should the dialplan look like:
>>>>>> 
>>>>>> 
>>>>>> <include>
>>>>>>   <extension name="public_did">
>>>>>>     <condition field="destination_number" expression="^(88888888)$">  
>>>>>> 	<action application="set" data="domain_name=$${domain}"/>
>>>>>> 	<action application="log" data="INFO domain is $${domain}"/>
>>>>>> 	<action application="set" data="callerID = $${caller_id_number}"/>	
>>>>>> 	<action application="log" data="INFO src number is $${callerID}"/>
>>>>>> <action application="respond" data="503"/>
>>>>>> 	<action application="bridge" data="{sip_auth_username=1003,sip_auth_password=R at D~o$121$@IIITD}sofia/external/1003 at 52.38.67.78 9769658019"/>
>>>>>> 	<!--<action application="respond" data="reponse_info" data="407"/> -->
>>>>>> 	
>>>>>> 	<!--<action application="transfer" data="5000 XML default"/> -->
>>>>>> 	
>>>>>> 	
>>>>>> 	  
>>>>>>     </condition>
>>>>>>   </extension>
>>>>>> </include>
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Regards,
>>>>>> Deepika
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services: 
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From colin.morelli at gmail.com  Sun Jun 12 23:07:33 2016
From: colin.morelli at gmail.com (Colin Morelli)
Date: Sun, 12 Jun 2016 19:07:33 +0000
Subject: [Freeswitch-users] Authentication of SIP phones against LDAP
	directory?
In-Reply-To: <F222D7AB-7728-42E7-A1BF-B1D761726FA5@haloprivacy.com>
References: <A6F2618F-2966-4C32-B49A-A35CC03A8164@haloprivacy.com>
	<CACOYK=jK6w8xzuTdgpPAto-Zd-W71H6np0R2xKFfyj1Qxoz=7g@mail.gmail.com>
	<F222D7AB-7728-42E7-A1BF-B1D761726FA5@haloprivacy.com>
Message-ID: <CAPtU-UoePmjXZELTBgVHZrzgFf0d8tRZmAiVUTwAL8kddFE=Yw@mail.gmail.com>

Mod_xml_curl is going to be your best bet for authenticating dynamically.
Build a simple HTTP service in your preferred language. LDAP Bindings are
widely available in basically every language.

With mod_xml_curl, FS will make an HTTP call to your application each time
someone tries to authenticate. You simply need to return a small XML
document back with their credentials.

Best,
Colin
On Sun, Jun 12, 2016 at 2:42 PM Kevin Long <kevin.long at haloprivacy.com>
wrote:

>
> Hi Stanislav,  thanks for the response.
>
> Even if I do need some separate attributes, plaintext or A1 hashed,  does
> the functionality exist in Freeswitch to do the authentication from LDAP
> from these attributes?
>
> My goal here is to run Freeswitch via Docker, because I intend to deploy
> it many, many times for small groups of users in an ephemeral way.  So the
> configuration files, including the users extensions etc,  would not be
> permanent and need to be generated on-the-fly when the instance is booted
> up.
>
>
> If there is another way people are doing this kind of thing,  I would love
> to know.
>
>
> Thanks again,
>
> Kevin Long
>
>
> > On Jun 12, 2016, at 7:18 AM, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
> >
> > there's a principal difference in how authentication works in SIP and
> LDAP:
> >
> > SIP is using challenge-response, so the server needs either the
> > cleartext password, or A1-hash.
> >
> > LDAP is primarily using salted hashes to store the passwords, so
> > there's no way to retrieve the cleartext password.
> >
> > So, the best you can do, is to have a separate LDAP attribute for the
> > SIP password, and keep clear text passwords in it. But then it comes
> > to the same problem as before, that the users have to maintain two
> > different passwords.
> >
> >
> >
> >
> > On Sun, Jun 12, 2016 at 1:27 AM, Kevin Long <kevin.long at haloprivacy.com>
> wrote:
> >>
> >>
> >> Hello,
> >>
> >>
> >> Can Freeswitch authenticate SIP phone logins from an LDAP directory ?
> >>
> >> Hoping to integrate this so my SIP users can use the same
> password/username they use for all other applications.
> >>
> >>
> >> Regards,
> >>
> >> Kevin Long
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From krice at freeswitch.org  Mon Jun 13 00:00:44 2016
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 12 Jun 2016 15:00:44 -0500
Subject: [Freeswitch-users] Authentication of SIP phones against LDAP
	directory?
In-Reply-To: <CAPtU-UoePmjXZELTBgVHZrzgFf0d8tRZmAiVUTwAL8kddFE=Yw@mail.gmail.com>
References: <A6F2618F-2966-4C32-B49A-A35CC03A8164@haloprivacy.com>
	<CACOYK=jK6w8xzuTdgpPAto-Zd-W71H6np0R2xKFfyj1Qxoz=7g@mail.gmail.com>
	<F222D7AB-7728-42E7-A1BF-B1D761726FA5@haloprivacy.com>
	<CAPtU-UoePmjXZELTBgVHZrzgFf0d8tRZmAiVUTwAL8kddFE=Yw@mail.gmail.com>
Message-ID: <64E5327C-5F4D-4946-9A20-97D817D91578@freeswitch.org>

https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/src/mod/directories/mod_ldap


Sent from my iPhone

> On Jun 12, 2016, at 2:07 PM, Colin Morelli <colin.morelli at gmail.com> wrote:
> 
> Mod_xml_curl is going to be your best bet for authenticating dynamically. Build a simple HTTP service in your preferred language. LDAP Bindings are widely available in basically every language.
> 
> With mod_xml_curl, FS will make an HTTP call to your application each time someone tries to authenticate. You simply need to return a small XML document back with their credentials.
> 
> Best,
> Colin
>> On Sun, Jun 12, 2016 at 2:42 PM Kevin Long <kevin.long at haloprivacy.com> wrote:
>> 
>> Hi Stanislav,  thanks for the response.
>> 
>> Even if I do need some separate attributes, plaintext or A1 hashed,  does the functionality exist in Freeswitch to do the authentication from LDAP from these attributes?
>> 
>> My goal here is to run Freeswitch via Docker, because I intend to deploy it many, many times for small groups of users in an ephemeral way.  So the configuration files, including the users extensions etc,  would not be permanent and need to be generated on-the-fly when the instance is booted up.
>> 
>> 
>> If there is another way people are doing this kind of thing,  I would love to know.
>> 
>> 
>> Thanks again,
>> 
>> Kevin Long
>> 
>> 
>> > On Jun 12, 2016, at 7:18 AM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
>> >
>> > there's a principal difference in how authentication works in SIP and LDAP:
>> >
>> > SIP is using challenge-response, so the server needs either the
>> > cleartext password, or A1-hash.
>> >
>> > LDAP is primarily using salted hashes to store the passwords, so
>> > there's no way to retrieve the cleartext password.
>> >
>> > So, the best you can do, is to have a separate LDAP attribute for the
>> > SIP password, and keep clear text passwords in it. But then it comes
>> > to the same problem as before, that the users have to maintain two
>> > different passwords.
>> >
>> >
>> >
>> >
>> > On Sun, Jun 12, 2016 at 1:27 AM, Kevin Long <kevin.long at haloprivacy.com> wrote:
>> >>
>> >>
>> >> Hello,
>> >>
>> >>
>> >> Can Freeswitch authenticate SIP phone logins from an LDAP directory ?
>> >>
>> >> Hoping to integrate this so my SIP users can use the same password/username they use for all other applications.
>> >>
>> >>
>> >> Regards,
>> >>
>> >> Kevin Long
>> >>
>> >>
>> >> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://confluence.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From kevin.long at haloprivacy.com  Mon Jun 13 00:20:08 2016
From: kevin.long at haloprivacy.com (Kevin Long)
Date: Sun, 12 Jun 2016 20:20:08 +0000
Subject: [Freeswitch-users] Authentication of SIP phones against
	LDAP	directory?
In-Reply-To: <CAPtU-UoePmjXZELTBgVHZrzgFf0d8tRZmAiVUTwAL8kddFE=Yw@mail.gmail.com>
References: <A6F2618F-2966-4C32-B49A-A35CC03A8164@haloprivacy.com>
	<CACOYK=jK6w8xzuTdgpPAto-Zd-W71H6np0R2xKFfyj1Qxoz=7g@mail.gmail.com>
	<F222D7AB-7728-42E7-A1BF-B1D761726FA5@haloprivacy.com>
	<CAPtU-UoePmjXZELTBgVHZrzgFf0d8tRZmAiVUTwAL8kddFE=Yw@mail.gmail.com>
Message-ID: <61FEBC17-4F57-4BC8-AC86-B02405A4EB1E@haloprivacy.com>


Thanks Colin,

That sounds like a workable idea and would come in handy for other dynamic parts of the configuration .

Cheers,

Kevin Long


On Jun 12, 2016, at 12:07 PM, Colin Morelli <colin.morelli at gmail.com<mailto:colin.morelli at gmail.com>> wrote:

Mod_xml_curl is going to be your best bet for authenticating dynamically. Build a simple HTTP service in your preferred language. LDAP Bindings are widely available in basically every language.

With mod_xml_curl, FS will make an HTTP call to your application each time someone tries to authenticate. You simply need to return a small XML document back with their credentials.

Best,
Colin
On Sun, Jun 12, 2016 at 2:42 PM Kevin Long <kevin.long at haloprivacy.com<mailto:kevin.long at haloprivacy.com>> wrote:

Hi Stanislav,  thanks for the response.

Even if I do need some separate attributes, plaintext or A1 hashed,  does the functionality exist in Freeswitch to do the authentication from LDAP from these attributes?

My goal here is to run Freeswitch via Docker, because I intend to deploy it many, many times for small groups of users in an ephemeral way.  So the configuration files, including the users extensions etc,  would not be permanent and need to be generated on-the-fly when the instance is booted up.


If there is another way people are doing this kind of thing,  I would love to know.


Thanks again,

Kevin Long


> On Jun 12, 2016, at 7:18 AM, Stanislav Sinyagin <ssinyagin at gmail.com<mailto:ssinyagin at gmail.com>> wrote:
>
> there's a principal difference in how authentication works in SIP and LDAP:
>
> SIP is using challenge-response, so the server needs either the
> cleartext password, or A1-hash.
>
> LDAP is primarily using salted hashes to store the passwords, so
> there's no way to retrieve the cleartext password.
>
> So, the best you can do, is to have a separate LDAP attribute for the
> SIP password, and keep clear text passwords in it. But then it comes
> to the same problem as before, that the users have to maintain two
> different passwords.
>
>
>
>
> On Sun, Jun 12, 2016 at 1:27 AM, Kevin Long <kevin.long at haloprivacy.com<mailto:kevin.long at haloprivacy.com>> wrote:
>>
>>
>> Hello,
>>
>>
>> Can Freeswitch authenticate SIP phone logins from an LDAP directory ?
>>
>> Hoping to integrate this so my SIP users can use the same password/username they use for all other applications.
>>
>>
>> Regards,
>>
>> Kevin Long
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/>
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org<http://www.freeswitch.org/>
>> http://confluence.freeswitch.org<http://confluence.freeswitch.org/>
>> http://www.cluecon.com<http://www.cluecon.com/>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org<http://www.freeswitch.org/>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/>
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org<http://www.freeswitch.org/>
> http://confluence.freeswitch.org<http://confluence.freeswitch.org/>
> http://www.cluecon.com<http://www.cluecon.com/>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org<http://www.freeswitch.org/>


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

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

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

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

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

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From brian at freeswitch.org  Mon Jun 13 03:59:53 2016
From: brian at freeswitch.org (Brian West)
Date: Sun, 12 Jun 2016 18:59:53 -0500
Subject: [Freeswitch-users] SHA-2 Implementation
In-Reply-To: <CAO=waFBp1ox46+Pe3hfsEw6gNWCCCiweC0P-9VEjrPRdMSWHCA@mail.gmail.com>
References: <CAO=waFBp1ox46+Pe3hfsEw6gNWCCCiweC0P-9VEjrPRdMSWHCA@mail.gmail.com>
Message-ID: <CAEJMVkBQoGKNVwV-xEOY3rMtgzZMXBAtUg0ynYo67WKrm8xkiw@mail.gmail.com>

This is a feature of the certificate you generate, not specifically
FreeSWITCH

On Friday, June 10, 2016, Abdul Hakeem <alhakeem at gmail.com> wrote:

> Hello,
>
> Any plans of moving FS from SHA-1 to SHA-2 crypto hash support ?
> I understand SHA-1 will be deprecated in Chrome by December, this will
> cause all calls to Verto to fail.
>
> Cheers,
> AH
>


-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com
https://www.gofundme.com/freeswitch_ubuntu

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

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From deepikay at iiitd.ac.in  Mon Jun 13 07:02:54 2016
From: deepikay at iiitd.ac.in (Deepika Yadav)
Date: Mon, 13 Jun 2016 08:32:54 +0530
Subject: [Freeswitch-users] Reject Inbound Call and Callback
In-Reply-To: <C4862E63-64CD-4DA9-A30A-99FFAF7F6D06@freeswitch.org>
References: <CAM99_Y=u+7gSogm1Lr-UaiAsSjYwVW0Av4xbzgh=MJuvoqdiYA@mail.gmail.com>
	<CAM99_Y=czCKWSmgFqfZZJ+68FkqA_z+X65BF2K18CHbgF2XGtw@mail.gmail.com>
	<5D156F9A-BEC7-433E-BF74-A6654C417DBD@freeswitch.org>
	<CAM99_Y=SVtSeQZVXGoCZ9fZMDfVthH3JJtVTTbN3OXxpgcQBOA@mail.gmail.com>
	<CAGVOjVU4VcoO=62kpTBxyopOO1z1Vq85kHENd9jL+p5fQtd7vw@mail.gmail.com>
	<CAM99_Ymvih91h0aT06PVodK72THPXyR9upJJTM_EqyfZrak2cg@mail.gmail.com>
	<C4862E63-64CD-4DA9-A30A-99FFAF7F6D06@freeswitch.org>
Message-ID: <CAM99_YnOrpxKgbwM14mMjuN0e1TwOf4AMcedte2puWPRg-qR3g@mail.gmail.com>

Thanks Ken,

I have figured it out, I used a Python module that reject the call by
hangup() and then created a new session for outgoing call.
On 13-Jun-2016 12:39 am, "Ken Rice" <krice at freeswitch.org> wrote:

> Ok let me try this again. Once the incoming call is hung up, the call is
> dead and anything processing on the channel (except the CDR) is done. No
> further commands in the dial-plan will execute.
>
>
> If you need pro help with this contact consulting at freeswitch.org and one
> of us pros can knock this out pretty fast for you
>
> Sent from my iPhone
>
> On Jun 12, 2016, at 12:01 PM, Deepika Yadav <deepikay at iiitd.ac.in> wrote:
>
> Even if I use Hangup command dialplan won't execute the statements after
> hangup.
>
> My basic requirement is when Freeswitch receives a call on an extension,
> it should reject it and then callback.
>
> On Sun, Jun 12, 2016 at 9:00 PM, Aqs Younas <aqsyounas at gmail.com> wrote:
>
>> You can hangup the call with reason.
>> https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_hangup
>> On Jun 12, 2016 7:36 PM, "Deepika Yadav" <deepikay at iiitd.ac.in> wrote:
>>
>>> Even in ESL, first I have to reject the call and start a new session.
>>>
>>> What is the command to reject a call in mod_python, I tried
>>> session.respond() that did not work.
>>>
>>>
>>> Python script :
>>>
>>> def handler(session, args):
>>>
>>>     namer = session.getVariable('caller_id_number')
>>>     freeswitch.consoleLog('info','Called by '+namer)
>>>     session.respond()
>>>     s = freeswitch.Session();
>>>     s.originate(session,"
>>> {sip_auth_username=1002,sip_auth_password=XXXXXXX}
>>> sofia/external/1002 at XX.XX.XX.XX <sofia/external/1002 at xx.xx.xx.xx>
>>>  9868789909")
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Sun, Jun 12, 2016 at 12:12 AM, Ken Rice <krice at freeswitch.org> wrote:
>>>
>>>> Look at your logs you'll notice that originate is not a dialplan
>>>> command and after you hang up on the incoming call the channel is dead. The
>>>> correct way to do this is with a script (probably ESl) that sees the
>>>> incoming call hangs it up and triggers the originate command
>>>>
>>>> Sent from my iPhone
>>>>
>>>> On Jun 11, 2016, at 1:26 PM, Deepika Yadav <deepikay at iiitd.ac.in>
>>>> wrote:
>>>>
>>>> sorry for the incomplete message, please ignore the previous message.
>>>>
>>>> I am creating the dialplan in this way:
>>>>
>>>> <include>
>>>>   <extension name="public_did">
>>>>     <condition field="destination_number" expression="^(88888888)$">
>>>> <action application="set" data="callerID = $${caller_id_number}"/>
>>>> <action application="log" data="INFO src number is $${callerID}"/>
>>>>         <action application="respond" data="503"/>
>>>> <action application="originate"
>>>> data="{sip_auth_username=1003,sip_auth_password=*******}sofia/external/
>>>> 1003@ <1003 at 52.38.67.78>X.X.X.X $callerID "/>
>>>>     </condition>
>>>>   </extension>
>>>> </include>
>>>>
>>>>
>>>> <extension name="show_prepare">
>>>> <condition field="destination_number"
>>>> expression="^(?:\+91|0|)(\d{10})$">
>>>>
>>>> <action application="bridge" data="sofia/gateway/MySIP/91$1"/>
>>>>       </condition>
>>>> </extension>
>>>>
>>>> After rejecting the call, new channel for outgoing call does not start
>>>> , also variable "callerID" is set to empty not capturing the source
>>>> number
>>>>
>>>> Regards,
>>>> Deepika
>>>>
>>>>
>>>>
>>>> On Sat, Jun 11, 2016 at 11:50 PM, Deepika Yadav <deepikay at iiitd.ac.in>
>>>> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> I want to reject an incoming call from a mobile number in the public
>>>>> context and callback to it
>>>>>
>>>>> how should the dialplan look like:
>>>>>
>>>>>
>>>>> <include>
>>>>>   <extension name="public_did">
>>>>>     <condition field="destination_number" expression="^(88888888)$">
>>>>> <action application="set" data="domain_name=$${domain}"/>
>>>>> <action application="log" data="INFO domain is $${domain}"/>
>>>>> <action application="set" data="callerID = $${caller_id_number}"/>
>>>>> <action application="log" data="INFO src number is $${callerID}"/>
>>>>> <action application="respond" data="503"/>
>>>>> <action application="bridge"
>>>>> data="{sip_auth_username=1003,sip_auth_password=R at D
>>>>> ~o$121$@IIITD}sofia/external/1003 at 52.38.67.78 9769658019"/>
>>>>> <!--<action application="respond" data="reponse_info" data="407"/> -->
>>>>> <!--<action application="transfer" data="5000 XML default"/> -->
>>>>>
>>>>>     </condition>
>>>>>   </extension>
>>>>> </include>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> Regards,
>>>>> Deepika
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://
>>>> lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From joel at gogii.net  Mon Jun 13 09:03:39 2016
From: joel at gogii.net (Joel Serrano)
Date: Sun, 12 Jun 2016 22:03:39 -0700
Subject: [Freeswitch-users] Reject Inbound Call and Callback
In-Reply-To: <CAM99_YnOrpxKgbwM14mMjuN0e1TwOf4AMcedte2puWPRg-qR3g@mail.gmail.com>
References: <CAM99_Y=u+7gSogm1Lr-UaiAsSjYwVW0Av4xbzgh=MJuvoqdiYA@mail.gmail.com>
	<CAM99_Y=czCKWSmgFqfZZJ+68FkqA_z+X65BF2K18CHbgF2XGtw@mail.gmail.com>
	<5D156F9A-BEC7-433E-BF74-A6654C417DBD@freeswitch.org>
	<CAM99_Y=SVtSeQZVXGoCZ9fZMDfVthH3JJtVTTbN3OXxpgcQBOA@mail.gmail.com>
	<CAGVOjVU4VcoO=62kpTBxyopOO1z1Vq85kHENd9jL+p5fQtd7vw@mail.gmail.com>
	<CAM99_Ymvih91h0aT06PVodK72THPXyR9upJJTM_EqyfZrak2cg@mail.gmail.com>
	<C4862E63-64CD-4DA9-A30A-99FFAF7F6D06@freeswitch.org>
	<CAM99_YnOrpxKgbwM14mMjuN0e1TwOf4AMcedte2puWPRg-qR3g@mail.gmail.com>
Message-ID: <CAMtXxQm5nnJ-wn-zZzpN4td3zqN+pL6fYg4AALsCuMMBoqwP=w@mail.gmail.com>

Hi Deepika,

Would you mind sharing how you did it? I'm kinda curious. Do you subscribe
to events through python and accordingly or do you run a python script from
the dialplan?

I'd love to see your python script as I normally use LUA and I want to give
python a try.

Thanks,
Joel.



On Sun, Jun 12, 2016 at 8:02 PM, Deepika Yadav <deepikay at iiitd.ac.in> wrote:

> Thanks Ken,
>
> I have figured it out, I used a Python module that reject the call by
> hangup() and then created a new session for outgoing call.
> On 13-Jun-2016 12:39 am, "Ken Rice" <krice at freeswitch.org> wrote:
>
>> Ok let me try this again. Once the incoming call is hung up, the call is
>> dead and anything processing on the channel (except the CDR) is done. No
>> further commands in the dial-plan will execute.
>>
>>
>> If you need pro help with this contact consulting at freeswitch.org and one
>> of us pros can knock this out pretty fast for you
>>
>> Sent from my iPhone
>>
>> On Jun 12, 2016, at 12:01 PM, Deepika Yadav <deepikay at iiitd.ac.in> wrote:
>>
>> Even if I use Hangup command dialplan won't execute the statements after
>> hangup.
>>
>> My basic requirement is when Freeswitch receives a call on an extension,
>> it should reject it and then callback.
>>
>> On Sun, Jun 12, 2016 at 9:00 PM, Aqs Younas <aqsyounas at gmail.com> wrote:
>>
>>> You can hangup the call with reason.
>>> https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_hangup
>>> On Jun 12, 2016 7:36 PM, "Deepika Yadav" <deepikay at iiitd.ac.in> wrote:
>>>
>>>> Even in ESL, first I have to reject the call and start a new session.
>>>>
>>>> What is the command to reject a call in mod_python, I tried
>>>> session.respond() that did not work.
>>>>
>>>>
>>>> Python script :
>>>>
>>>> def handler(session, args):
>>>>
>>>>     namer = session.getVariable('caller_id_number')
>>>>     freeswitch.consoleLog('info','Called by '+namer)
>>>>     session.respond()
>>>>     s = freeswitch.Session();
>>>>     s.originate(session,"
>>>> {sip_auth_username=1002,sip_auth_password=XXXXXXX}
>>>> sofia/external/1002 at XX.XX.XX.XX <sofia/external/1002 at xx.xx.xx.xx>
>>>>  9868789909")
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Sun, Jun 12, 2016 at 12:12 AM, Ken Rice <krice at freeswitch.org>
>>>> wrote:
>>>>
>>>>> Look at your logs you'll notice that originate is not a dialplan
>>>>> command and after you hang up on the incoming call the channel is dead. The
>>>>> correct way to do this is with a script (probably ESl) that sees the
>>>>> incoming call hangs it up and triggers the originate command
>>>>>
>>>>> Sent from my iPhone
>>>>>
>>>>> On Jun 11, 2016, at 1:26 PM, Deepika Yadav <deepikay at iiitd.ac.in>
>>>>> wrote:
>>>>>
>>>>> sorry for the incomplete message, please ignore the previous message.
>>>>>
>>>>> I am creating the dialplan in this way:
>>>>>
>>>>> <include>
>>>>>   <extension name="public_did">
>>>>>     <condition field="destination_number" expression="^(88888888)$">
>>>>> <action application="set" data="callerID = $${caller_id_number}"/>
>>>>> <action application="log" data="INFO src number is $${callerID}"/>
>>>>>         <action application="respond" data="503"/>
>>>>> <action application="originate"
>>>>> data="{sip_auth_username=1003,sip_auth_password=*******}sofia/external/
>>>>> 1003@ <1003 at 52.38.67.78>X.X.X.X $callerID "/>
>>>>>     </condition>
>>>>>   </extension>
>>>>> </include>
>>>>>
>>>>>
>>>>> <extension name="show_prepare">
>>>>> <condition field="destination_number"
>>>>> expression="^(?:\+91|0|)(\d{10})$">
>>>>>
>>>>> <action application="bridge" data="sofia/gateway/MySIP/91$1"/>
>>>>>       </condition>
>>>>> </extension>
>>>>>
>>>>> After rejecting the call, new channel for outgoing call does not start
>>>>> , also variable "callerID" is set to empty not capturing the source
>>>>> number
>>>>>
>>>>> Regards,
>>>>> Deepika
>>>>>
>>>>>
>>>>>
>>>>> On Sat, Jun 11, 2016 at 11:50 PM, Deepika Yadav <deepikay at iiitd.ac.in>
>>>>> wrote:
>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> I want to reject an incoming call from a mobile number in the public
>>>>>> context and callback to it
>>>>>>
>>>>>> how should the dialplan look like:
>>>>>>
>>>>>>
>>>>>> <include>
>>>>>>   <extension name="public_did">
>>>>>>     <condition field="destination_number" expression="^(88888888)$">
>>>>>> <action application="set" data="domain_name=$${domain}"/>
>>>>>> <action application="log" data="INFO domain is $${domain}"/>
>>>>>> <action application="set" data="callerID = $${caller_id_number}"/>
>>>>>> <action application="log" data="INFO src number is $${callerID}"/>
>>>>>> <action application="respond" data="503"/>
>>>>>> <action application="bridge"
>>>>>> data="{sip_auth_username=1003,sip_auth_password=R at D
>>>>>> ~o$121$@IIITD}sofia/external/1003 at 52.38.67.78 9769658019"/>
>>>>>> <!--<action application="respond" data="reponse_info" data="407"/> -->
>>>>>> <!--<action application="transfer" data="5000 XML default"/> -->
>>>>>>
>>>>>>     </condition>
>>>>>>   </extension>
>>>>>> </include>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> Regards,
>>>>>> Deepika
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://
>>>>> lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From flokrrr at gmail.com  Mon Jun 13 11:29:21 2016
From: flokrrr at gmail.com (Florent Krieg)
Date: Mon, 13 Jun 2016 09:29:21 +0200
Subject: [Freeswitch-users] Channel/session variables availability on
	execute_on_sip_reinvite
In-Reply-To: <9CA4C4E8-9FE0-4413-BA60-7A303EF4AFA6@jerris.com>
References: <CAABKsuRM0A-k5kMn+gsHHuQJYKVc6mT_mcqEdGDnt2MopL6t0g@mail.gmail.com>
	<9CA4C4E8-9FE0-4413-BA60-7A303EF4AFA6@jerris.com>
Message-ID: <CAABKsuRwRXW8Dk16UnuwzRCT_LxNcGN2UAZsGoRWWSzi=tR2Cg@mail.gmail.com>

Hi Michael,

The raw sdp is actually available, but the thing is that it doesn't look
like the one I see in the re-invite when I capture on the host.
It looks as if it has already been processed by the SIP stack, depending on
the media configuration of the sip profile and/or the current channel.
Is it possible?

Florent

2016-06-10 19:14 GMT+02:00 Michael Jerris <mike at jerris.com>:

> I don't think we have the sdp parsed and processed at the point we get
> that callback.  Check the variables to see if the raw sdp is available in
> the var yet or not.
>
> > On Jun 10, 2016, at 4:40 AM, Florent Krieg <flokrrr at gmail.com> wrote:
> >
> > Hello everybody,
> >
> > My use case is pretty simple: I would like to react on SIP Re-INVITE
> without sdp (empty body/content-length 0) so I have put an
> execute_on_sip_reinvite instruction in my bridge string.
> >
> > I have tried both to call the 'info' app and to call a custom shell
> system script, passing the ${uuid} var and calling fs_cli uuid_dump $uuid,
> but unfortunately it seems that when the SIP message reaches the
> application it has been processed by the Sofia stack and I can't detect if
> there is an SDP or not (using sdp variables, or any other actually).
> > Sometimes it gives me 0.0.0.0 as the remote_media_ip, sometimes it just
> builds an SDP based on the previous one.
> > I have also tried to modify the re-INVITE by adding custom headers or
> putting information in any
> > mandatory SIP header (like, say, Via), but these updates doesn't seem to
> be available via any channel variable.
> >
> > Is there any FS configuration parameter I could change to be able to
> access this information?
> > Is there any other way I could do what I want?
> >
> >
> > Thanks in advance if you have any clue :)
> >
> > Florent
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From deepikay at iiitd.ac.in  Mon Jun 13 13:56:30 2016
From: deepikay at iiitd.ac.in (Deepika Yadav)
Date: Mon, 13 Jun 2016 15:26:30 +0530
Subject: [Freeswitch-users] Reject Inbound Call and Callback
In-Reply-To: <CAMtXxQm5nnJ-wn-zZzpN4td3zqN+pL6fYg4AALsCuMMBoqwP=w@mail.gmail.com>
References: <CAM99_Y=u+7gSogm1Lr-UaiAsSjYwVW0Av4xbzgh=MJuvoqdiYA@mail.gmail.com>
	<CAM99_Y=czCKWSmgFqfZZJ+68FkqA_z+X65BF2K18CHbgF2XGtw@mail.gmail.com>
	<5D156F9A-BEC7-433E-BF74-A6654C417DBD@freeswitch.org>
	<CAM99_Y=SVtSeQZVXGoCZ9fZMDfVthH3JJtVTTbN3OXxpgcQBOA@mail.gmail.com>
	<CAGVOjVU4VcoO=62kpTBxyopOO1z1Vq85kHENd9jL+p5fQtd7vw@mail.gmail.com>
	<CAM99_Ymvih91h0aT06PVodK72THPXyR9upJJTM_EqyfZrak2cg@mail.gmail.com>
	<C4862E63-64CD-4DA9-A30A-99FFAF7F6D06@freeswitch.org>
	<CAM99_YnOrpxKgbwM14mMjuN0e1TwOf4AMcedte2puWPRg-qR3g@mail.gmail.com>
	<CAMtXxQm5nnJ-wn-zZzpN4td3zqN+pL6fYg4AALsCuMMBoqwP=w@mail.gmail.com>
Message-ID: <CAM99_YnhdHeE2KrKKN7Zoy=ECd1ACtXBRuEvLiQbAG+LKFrXjg@mail.gmail.com>

You can follow the mod_python Wiki :

https://wiki.freeswitch.org/wiki/Mod_python

Inside the handler()

session.hangup()
new_api_obj = API()
command_string = "originate
{sip_auth_username=1002,sip_auth_password=XXXXXXX}sofia/external/1002 at XX.XX.XX.XX
"+dialled_number
new_api_obj.executeString(command_string)

Here, it creates a new session after previous session is hungup completely
and makes an outbound call


I want play my own IVR menu when dialling to the dialled_number, to do some
actions on DTMF inputs.
So, I use the command :

command_string = "sofia/gateway/MySIP/"+callerid+" 4449"

Here, it does not create  a new session and report as "USER_BUSY", since it
did not hungup in case before creating a new channel





On Mon, Jun 13, 2016 at 10:33 AM, Joel Serrano <joel at gogii.net> wrote:

> Hi Deepika,
>
> Would you mind sharing how you did it? I'm kinda curious. Do you subscribe
> to events through python and accordingly or do you run a python script from
> the dialplan?
>
> I'd love to see your python script as I normally use LUA and I want to
> give python a try.
>
> Thanks,
> Joel.
>
>
>
> On Sun, Jun 12, 2016 at 8:02 PM, Deepika Yadav <deepikay at iiitd.ac.in>
> wrote:
>
>> Thanks Ken,
>>
>> I have figured it out, I used a Python module that reject the call by
>> hangup() and then created a new session for outgoing call.
>> On 13-Jun-2016 12:39 am, "Ken Rice" <krice at freeswitch.org> wrote:
>>
>>> Ok let me try this again. Once the incoming call is hung up, the call is
>>> dead and anything processing on the channel (except the CDR) is done. No
>>> further commands in the dial-plan will execute.
>>>
>>>
>>> If you need pro help with this contact consulting at freeswitch.org and
>>> one of us pros can knock this out pretty fast for you
>>>
>>> Sent from my iPhone
>>>
>>> On Jun 12, 2016, at 12:01 PM, Deepika Yadav <deepikay at iiitd.ac.in>
>>> wrote:
>>>
>>> Even if I use Hangup command dialplan won't execute the statements after
>>> hangup.
>>>
>>> My basic requirement is when Freeswitch receives a call on an extension,
>>> it should reject it and then callback.
>>>
>>> On Sun, Jun 12, 2016 at 9:00 PM, Aqs Younas <aqsyounas at gmail.com> wrote:
>>>
>>>> You can hangup the call with reason.
>>>> https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_hangup
>>>> On Jun 12, 2016 7:36 PM, "Deepika Yadav" <deepikay at iiitd.ac.in> wrote:
>>>>
>>>>> Even in ESL, first I have to reject the call and start a new session.
>>>>>
>>>>> What is the command to reject a call in mod_python, I tried
>>>>> session.respond() that did not work.
>>>>>
>>>>>
>>>>> Python script :
>>>>>
>>>>> def handler(session, args):
>>>>>
>>>>>     namer = session.getVariable('caller_id_number')
>>>>>     freeswitch.consoleLog('info','Called by '+namer)
>>>>>     session.respond()
>>>>>     s = freeswitch.Session();
>>>>>     s.originate(session,"
>>>>> {sip_auth_username=1002,sip_auth_password=XXXXXXX}
>>>>> sofia/external/1002 at XX.XX.XX.XX <sofia/external/1002 at xx.xx.xx.xx>
>>>>>  9868789909")
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Sun, Jun 12, 2016 at 12:12 AM, Ken Rice <krice at freeswitch.org>
>>>>> wrote:
>>>>>
>>>>>> Look at your logs you'll notice that originate is not a dialplan
>>>>>> command and after you hang up on the incoming call the channel is dead. The
>>>>>> correct way to do this is with a script (probably ESl) that sees the
>>>>>> incoming call hangs it up and triggers the originate command
>>>>>>
>>>>>> Sent from my iPhone
>>>>>>
>>>>>> On Jun 11, 2016, at 1:26 PM, Deepika Yadav <deepikay at iiitd.ac.in>
>>>>>> wrote:
>>>>>>
>>>>>> sorry for the incomplete message, please ignore the previous message.
>>>>>>
>>>>>> I am creating the dialplan in this way:
>>>>>>
>>>>>> <include>
>>>>>>   <extension name="public_did">
>>>>>>     <condition field="destination_number" expression="^(88888888)$">
>>>>>> <action application="set" data="callerID = $${caller_id_number}"/>
>>>>>> <action application="log" data="INFO src number is $${callerID}"/>
>>>>>>         <action application="respond" data="503"/>
>>>>>> <action application="originate"
>>>>>> data="{sip_auth_username=1003,sip_auth_password=*******}sofia/external/
>>>>>> 1003@ <1003 at 52.38.67.78>X.X.X.X $callerID "/>
>>>>>>     </condition>
>>>>>>   </extension>
>>>>>> </include>
>>>>>>
>>>>>>
>>>>>> <extension name="show_prepare">
>>>>>> <condition field="destination_number"
>>>>>> expression="^(?:\+91|0|)(\d{10})$">
>>>>>>
>>>>>> <action application="bridge" data="sofia/gateway/MySIP/91$1"/>
>>>>>>       </condition>
>>>>>> </extension>
>>>>>>
>>>>>> After rejecting the call, new channel for outgoing call does not
>>>>>> start , also variable "callerID" is set to empty not capturing the
>>>>>> source number
>>>>>>
>>>>>> Regards,
>>>>>> Deepika
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Sat, Jun 11, 2016 at 11:50 PM, Deepika Yadav <deepikay at iiitd.ac.in
>>>>>> > wrote:
>>>>>>
>>>>>>> Hi,
>>>>>>>
>>>>>>> I want to reject an incoming call from a mobile number in the public
>>>>>>> context and callback to it
>>>>>>>
>>>>>>> how should the dialplan look like:
>>>>>>>
>>>>>>>
>>>>>>> <include>
>>>>>>>   <extension name="public_did">
>>>>>>>     <condition field="destination_number" expression="^(88888888)$">
>>>>>>>
>>>>>>> <action application="set" data="domain_name=$${domain}"/>
>>>>>>> <action application="log" data="INFO domain is $${domain}"/>
>>>>>>> <action application="set" data="callerID = $${caller_id_number}"/>
>>>>>>> <action application="log" data="INFO src number is $${callerID}"/>
>>>>>>> <action application="respond" data="503"/>
>>>>>>> <action application="bridge"
>>>>>>> data="{sip_auth_username=1003,sip_auth_password=R at D
>>>>>>> ~o$121$@IIITD}sofia/external/1003 at 52.38.67.78 9769658019"/>
>>>>>>> <!--<action application="respond" data="reponse_info" data="407"/>
>>>>>>> -->
>>>>>>> <!--<action application="transfer" data="5000 XML default"/> -->
>>>>>>>
>>>>>>>     </condition>
>>>>>>>   </extension>
>>>>>>> </include>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Regards,
>>>>>>> Deepika
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://
>>>>>> lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lists at kavun.ch  Mon Jun 13 15:34:46 2016
From: lists at kavun.ch (Emrah)
Date: Mon, 13 Jun 2016 13:34:46 +0200
Subject: [Freeswitch-users] distributed presence (again)
In-Reply-To: <5B0BFA80-9BA4-46BD-AD70-5170F48D1CE8@beachdognet.com>
References: <5B0BFA80-9BA4-46BD-AD70-5170F48D1CE8@beachdognet.com>
Message-ID: <79B2BC2C-4B26-43F5-A46C-9EC961D0F704@kavun.ch>

Thanks for this post Dave. Likewise I?m looking for a solution. Looks like DB support is all there already. Let me know how I can help.
If we are missing something and distributed presence is already functional, some info would be appreciated.
I cannot share the same hostname because each server must keep its own registrations in order for my distributed setup to work.

Emrah
> On Jun 10, 2016, at 4:21 AM, Dave Horton <daveh at beachdognet.com> wrote:
> 
> I?ve seen a few recent posts on distributed presence work, and I am running into the same need.
> Like the other?s who have posted, I can?t use the same hostname across servers.
> In my case, this is because in order to make call transfer scenarios work I need
> to redirect calls sometimes to specific servers, and unique hostnames are required for that.
> 
> Is there another solution for this?  
> 
> If not, I?d be interested in working on a solution, as long as there was a view towards
> incorporating it in the mainstream code at some point (i.e., I wouldn't want to have to 
> keep patching releases going forward).
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From matt at supportedbusiness.com  Mon Jun 13 16:06:11 2016
From: matt at supportedbusiness.com (Matt Broad)
Date: Mon, 13 Jun 2016 13:06:11 +0100
Subject: [Freeswitch-users] add modules when installing from the latest
	branch
In-Reply-To: <B670C6AB-D0AD-4A17-AAF2-F8377F88C43D@freeswitch.org>
References: <CAEeV61jv_rX-ebF6O4gM3=DG3jEDdMaBBovYtj_2QAwzozPJrg@mail.gmail.com>
	<CAF7jizJ7yXS5a0QhFm5uY1vZttcmO5HoM3sssCAqLY1NJyO37g@mail.gmail.com>
	<CAEeV61g12Ar2euivZQb+3NkTB7mRO7keRe12AN7kp9J8Do3xZw@mail.gmail.com>
	<007501d1c0d2$2febc220$8fc34660$@freeswitch.org>
	<CAEeV61h+5Gy2Wo16jOEze2qPGmrD8q8pb3q01wTViAv2Sz+Z7A@mail.gmail.com>
	<B670C6AB-D0AD-4A17-AAF2-F8377F88C43D@freeswitch.org>
Message-ID: <CAEeV61ihw-aHUJxGWqNmoW-VqBhsALUwWe1VhGg2n-Y6Sn42MA@mail.gmail.com>

Hi,

I'm trying to install freetdm for a sangoma a200 board, but cannot see the
freetdm module using the apt-get method.  Any ideas how to add this?

thanks
Matt

On 8 June 2016 at 23:59, Ken Rice <krice at freeswitch.org> wrote:

> With this method of install everything is in an fhs compliant location.
> You can mod the commandline args in he systemd files tho
>
> Sent from my iPhone
>
> On Jun 8, 2016, at 3:21 PM, Matt Broad <matt at supportedbusiness.com> wrote:
>
> thanks Ken, just another quick question on this install method; how would
> I go about changing the default locations for the scripts?
>
> I would ordinarily use the following: ./configure --with-scriptdir=
> /etc/freeswitch/scripts/
>
>
>
> thanks
> Matt
>
> On 7 June 2016 at 16:35, Ken Rice <krice at freeswitch.org> wrote:
>
>> Easier is probably just apt-cache search freeswitch or apt-cache search
>> freeswitch|grep mongo or something like that
>>
>>
>>
>> Once you do that you can apt-get install the module you want and then you
>> just need to enable it in your
>> /etc/freeswitch/autoload_configs/modules.conf.xml
>>
>>
>>
>> The entire package system is designed so that you can use
>> https://freeswitch.org/stash/projects/FS/repos/freeswitch-config-skel/browse
>> to generate your own configuration package with deps on the modules that
>> you want to include and just install that and bam you?re good
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Matt Broad
>> *Sent:* Tuesday, June 7, 2016 5:50 AM
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* Re: [Freeswitch-users] add modules when installing from the
>> latest branch
>>
>>
>>
>> perfect!
>>
>>
>>
>> thanks for the quick response.  Is this listed on the confluence site
>> anywhere?  I am happy to add this if given access
>>
>>
>>
>> thanks
>>
>> Matt
>>
>>
>>
>> On 7 June 2016 at 11:35, Benjamin Cropley <benjamin.cropley at gmail.com>
>> wrote:
>>
>> Say you wanted to install mod_mongo..
>>
>>
>>
>> you can find the packages here
>> http://files.freeswitch.org/repo/deb/freeswitch-1.6/dists/jessie/main/binary-amd64/Packages
>>
>>
>>
>> apt-get install -y freeswitch-mod-mongo
>>
>>
>>
>> then edit the modules conf as you would have done previously
>>
>>
>>
>> On Tue, Jun 7, 2016 at 11:20 AM, Matt Broad <matt at supportedbusiness.com>
>> wrote:
>>
>> Hi,
>>
>>
>>
>> this may be a really stupid question but here goes:
>>
>>
>>
>> I have upgraded a server to debian 8 and have installed freeswitch using
>> the latest branch
>>
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie#Debian8Jessie-latestreleasebranch
>> :
>>
>>
>>
>> My question is how do I activate other modules?  In previous versions I
>> would have amended the moudules conf file and re-compile.  I can't find the
>> modules file in this install
>>
>>
>>
>>
>>
>> thanks
>>
>> Matt
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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From daveh at beachdognet.com  Mon Jun 13 16:09:07 2016
From: daveh at beachdognet.com (Dave Horton)
Date: Mon, 13 Jun 2016 08:09:07 -0400
Subject: [Freeswitch-users] distributed presence (again)
In-Reply-To: <79B2BC2C-4B26-43F5-A46C-9EC961D0F704@kavun.ch>
References: <5B0BFA80-9BA4-46BD-AD70-5170F48D1CE8@beachdognet.com>
	<79B2BC2C-4B26-43F5-A46C-9EC961D0F704@kavun.ch>
Message-ID: <28106D4F-2A5F-4EFD-993A-83207A1D650B@beachdognet.com>

Wouldn?t it make sense for their to be a concept of a ?presence realm? which is different than a hostname (i,e, a grouping of multiple hosts)?
On Jun 13, 2016, at 7:34 AM, Emrah <lists at kavun.ch> wrote:

Thanks for this post Dave. Likewise I?m looking for a solution. Looks like DB support is all there already. Let me know how I can help.
If we are missing something and distributed presence is already functional, some info would be appreciated.
I cannot share the same hostname because each server must keep its own registrations in order for my distributed setup to work.

Emrah
> On Jun 10, 2016, at 4:21 AM, Dave Horton <daveh at beachdognet.com> wrote:
> 
> I?ve seen a few recent posts on distributed presence work, and I am running into the same need.
> Like the other?s who have posted, I can?t use the same hostname across servers.
> In my case, this is because in order to make call transfer scenarios work I need
> to redirect calls sometimes to specific servers, and unique hostnames are required for that.
> 
> Is there another solution for this?  
> 
> If not, I?d be interested in working on a solution, as long as there was a view towards
> incorporating it in the mainstream code at some point (i.e., I wouldn't want to have to 
> keep patching releases going forward).
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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

Official FreeSWITCH Sites
http://www.freeswitch.org
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From mike at jerris.com  Mon Jun 13 19:15:45 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 13 Jun 2016 11:15:45 -0400
Subject: [Freeswitch-users] Channel/session variables availability on
	execute_on_sip_reinvite
In-Reply-To: <CAABKsuRwRXW8Dk16UnuwzRCT_LxNcGN2UAZsGoRWWSzi=tR2Cg@mail.gmail.com>
References: <CAABKsuRM0A-k5kMn+gsHHuQJYKVc6mT_mcqEdGDnt2MopL6t0g@mail.gmail.com>
	<9CA4C4E8-9FE0-4413-BA60-7A303EF4AFA6@jerris.com>
	<CAABKsuRwRXW8Dk16UnuwzRCT_LxNcGN2UAZsGoRWWSzi=tR2Cg@mail.gmail.com>
Message-ID: <1CDCFEC3-D152-4845-AE80-77B044D6A4A2@jerris.com>

That seems strange to me... but I would need to dig to confirm.

> On Jun 13, 2016, at 3:29 AM, Florent Krieg <flokrrr at gmail.com> wrote:
> 
> Hi Michael,
> 
> The raw sdp is actually available, but the thing is that it doesn't look like the one I see in the re-invite when I capture on the host.
> It looks as if it has already been processed by the SIP stack, depending on the media configuration of the sip profile and/or the current channel.
> Is it possible?
> 
> Florent
> 
> 2016-06-10 19:14 GMT+02:00 Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>>:
> I don't think we have the sdp parsed and processed at the point we get that callback.  Check the variables to see if the raw sdp is available in the var yet or not.
> 
> > On Jun 10, 2016, at 4:40 AM, Florent Krieg <flokrrr at gmail.com <mailto:flokrrr at gmail.com>> wrote:
> >
> > Hello everybody,
> >
> > My use case is pretty simple: I would like to react on SIP Re-INVITE without sdp (empty body/content-length 0) so I have put an execute_on_sip_reinvite instruction in my bridge string.
> >
> > I have tried both to call the 'info' app and to call a custom shell system script, passing the ${uuid} var and calling fs_cli uuid_dump $uuid, but unfortunately it seems that when the SIP message reaches the application it has been processed by the Sofia stack and I can't detect if there is an SDP or not (using sdp variables, or any other actually).
> > Sometimes it gives me 0.0.0.0 as the remote_media_ip, sometimes it just builds an SDP based on the previous one.
> > I have also tried to modify the re-INVITE by adding custom headers or putting information in any
> > mandatory SIP header (like, say, Via), but these updates doesn't seem to be available via any channel variable.
> >
> > Is there any FS configuration parameter I could change to be able to access this information?
> > Is there any other way I could do what I want?
> >
> >
> > Thanks in advance if you have any clue :)
> >
> > Florent
> 

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From mirkobrankovic at gmail.com  Mon Jun 13 15:28:17 2016
From: mirkobrankovic at gmail.com (Mirko Brankovic)
Date: Mon, 13 Jun 2016 13:28:17 +0200
Subject: [Freeswitch-users] ReInvte in bypass/proxy mode | a > a+v
In-Reply-To: <CAHtxdDckjrTpjQmyhcJCZKfOnSD6Ar=4HzJpZTMx7KHTRuZYyA@mail.gmail.com>
References: <CAND18T0Yh6w1paAfAyrv5SHhM+7WrWhMyh257HGRg5T8FQo=nw@mail.gmail.com>
	<CAHtxdDckjrTpjQmyhcJCZKfOnSD6Ar=4HzJpZTMx7KHTRuZYyA@mail.gmail.com>
Message-ID: <CAND18T0RCAtApkxpxyYzZxdiKjFt-K3Khz1yj8tpNPN9nu4PJA@mail.gmail.com>

Hi Sergey,

I have recently came accross this tread:
http://lists.freeswitch.org/pipermail/freeswitch-users/2016-January/118069.html

So i'm not sure if there is a feature request to support this or not (can't
seem to find it).

Regards,
mirko

On Wed, Jun 1, 2016 at 9:59 PM, Sergey Safarov <s.safarov at gmail.com> wrote:

> Look at FS-9203 <https://freeswitch.org/jira/browse/FS-9203> and FS-9206
> <https://freeswitch.org/jira/browse/FS-9206>
>
>
> ??, 1 ???. 2016 ?. ? 19:12, Mirko Brankovic <mirkobrankovic at gmail.com>:
>
>> Hi,
>>
>> Can someone help me figure out the problem i am hitting.
>>
>> Im trying to upgrade call from audio to audio + video (opus 48k and vp8-
>> but this one is only in passtrough mode).
>>
>> So set up is browser (sip.js/node.js) >> kamailio wss >> freswitch
>> (bypass mode bridge without answer) >> kamailio >> browser (node.js)
>>
>> Problem is that freeswitch receives reInvite on existing audio call with
>> new sdp containing also video codecs, but directly from client so no proxy
>> (kamailio) in path, and exiting invite towards kamailio for outgoing leg is
>> missing video in a=group BUNDLE, so only audio and a=sendrcv is also being
>> stripped (that is the difference to 1 on 1 video call).
>>
>> I understood that bypass mode shouldn't change/touch sdp.
>> Also passtrough codec shouldn't be the problem i guess since both clients
>> can decode it.
>>
>> Any help is appreciated.
>> Thanks,
>> Mirko
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Regards,
Mirko
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From turja_cse06 at yahoo.com  Mon Jun 13 16:14:09 2016
From: turja_cse06 at yahoo.com (turja)
Date: Mon, 13 Jun 2016 05:14:09 -0700 (MST)
Subject: [Freeswitch-users] Need help on call bridge !!
Message-ID: <1465820049600-7596223.post@n2.nabble.com>

Hello
i am new user of Freeswitch. i am trying to write a dial plan that when i
call to 1234(callee) first call bridge to  1233(another number) after 1233
hangup the call bridge 1234(callee) ....but can not ...after hangup from
1233(another number)the call hangs up not bridging with 1234(callee).....
how to do that in freeswitch.
Thanks for help !

Turja



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Need-help-on-call-bridge-tp7596223.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From matthew at brightfire.net  Mon Jun 13 19:59:11 2016
From: matthew at brightfire.net (Matthew)
Date: Mon, 13 Jun 2016 10:59:11 -0500
Subject: [Freeswitch-users] Problems with wait_for_silence API call
In-Reply-To: <0B49608D-083E-4F32-96C9-176FC4E5A441@jerris.com>
References: <fbd83d1b-6309-e170-25f4-a62f73428802@brightfire.net>
	<4eb0ca23-886c-72b8-da7e-a12cc78329e5@brightfire.net>
	<0B49608D-083E-4F32-96C9-176FC4E5A441@jerris.com>
Message-ID: <e11c079e-93dc-559c-86e4-2a8d76ccd977@brightfire.net>

Hi Michael,

Thanks for the info. Pull request #881 has been submitted.

-Matthew

On 6/9/2016 5:00 PM, Michael Jerris wrote:
> Can you please submit a pull request for these changes so they can be reviewed?
>
>> On Jun 9, 2016, at 5:14 PM, Matthew <matthew at brightfire.net> wrote:
>>
>> Hello All,
>>
>> After poking around a bit more I realized is that wait_for_silence was
>> never actually waiting for silence. It needs to detect non-silence
>> before it will ever try to detect silence. That makes it quite useless
>> if you call the function from a point where you can't guarantee that
>> audio is being received :/ What I did was broke up the function into two
>> smaller functions ...
>>
>> detect_audio <threshold> <audio_hits> <timeout_ms> [<file>]
>> detect_silence <threshold> <silence_hits> <timeout_ms> [<file>]
>>
>> This allowed me the flexibility I needed to build an outbound dial
>> solution the way I wanted to. For example, I can now do ...
>>
>>    amd_detect = "person"
>>    dialout_session:execute( "voice_start" )
>>    -- wait up to 3 seconds for 5 frames of audio above threshold 300
>>    dialout_session:execute( "detect_audio", "300 5 3000" )
>>    detect_audio_timeout = dialout_session:getVariable(
>> "detect_audio_timeout" )
>>    dialout_session:consoleLog( "INFO", "dialout: detect_audio_timeout =
>> " .. detect_audio_timeout )
>>    -- if audio was detected, wait up to 30 seconds for 30 frames of
>> silence below threshold 300
>>    if detect_audio_timeout == "false" then
>>      dialout_session:execute( "detect_silence", "300 30 30000" )
>>    end
>>    dialout_session:execute( "voice_stop" )
>>    if( dialout_session:getVariable( "amd_status" ) ~= nil ) then
>>      amd_detect = dialout_session:getVariable( "amd_status" )
>>    end
>>    dialout_session:consoleLog( "INFO", "dialout: amd_detect = " ..
>> amd_detect )
>>    if amd_detect == "machine" then
>>      ...
>>    else
>>     ...
>>    end
>>
>> I have patches based on a current git repo. Should I submit those
>> somewhere for inclusion upstream for other people to use or should I
>> just maintain these locally?
>>
>> Thanks,
>>
>> -Matthew
>>
>> On 6/8/2016 4:05 PM, Matthew wrote:
>>> Hello All,
>>>
>>> I've having a problem with the wait_for_silence API call in lua. When I
>>> attempt to orginate a call and then check for silence on the line, I'm
>>> getting unpredictable results. Sometimes it seems to work, but more
>>> times than not it times out ( switch_ivr_wait_for_silence: TIMEOUT 0 ).
>>> To test this, I answer the call, speak a short greeting and then mute
>>> the handset. In other words, I know the script should be detecting
>>> silence on the other end of the call. Here is a good portion of the test
>>> script I'm using ...
>>>
>>> local cid_name   = argv[1]
>>> local cid_number = argv[2]
>>> local gw_string  = argv[3]
>>> local queue_ext  = argv[4]
>>>
>>> dialout_string =
>>>     "{origination_caller_id_name='" .. cid_name .. "'," ..
>>>     "origination_caller_id_number=" .. cid_number .. "," ..
>>>     "ignore_early_media=true}" .. gw_string
>>>
>>> dialout_session = freeswitch.Session( dialout_string )
>>>
>>> while dialout_session:ready() and not dialout_session:answered() do
>>>     freeswitch.msleep( 250 )
>>> end
>>>
>>> if dialout_session:ready() and dialout_session:answered() then
>>>
>>>     dialout_session:consoleLog( "INFO", "dialout: connected to " ..
>>> gw_string ) )
>>>
>>>     dialout_session:execute( "voice_start" )
>>>     dialout_session:sleep( 2000 )
>>>     dialout_session:execute( "wait_for_silence", "300 30 15 15000" )
>>>     dialout_session:execute( "voice_stop" )
>>>
>>>     dialout_session:consoleLog( "INFO", "dialout: amd_detect = " ..
>>> amd_detect )
>>>
>>> end
>>>
>>> And here is an example of how I would invoke the script ...
>>>
>>> luarun dialout.lua TEST 1234567890 sofia/gateway/twilio/+1234567890 7000
>>>
>>> Any ideas why the wait_for_silence API call would be misbehaving? Any
>>> suggestions on how I could debug this problem further?
>>>
>>> Thanks,
>>>
>>> -Matthew
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From joel at gogii.net  Mon Jun 13 20:56:45 2016
From: joel at gogii.net (Joel Serrano)
Date: Mon, 13 Jun 2016 09:56:45 -0700
Subject: [Freeswitch-users] Reject Inbound Call and Callback
In-Reply-To: <CAM99_YnhdHeE2KrKKN7Zoy=ECd1ACtXBRuEvLiQbAG+LKFrXjg@mail.gmail.com>
References: <CAM99_Y=u+7gSogm1Lr-UaiAsSjYwVW0Av4xbzgh=MJuvoqdiYA@mail.gmail.com>
	<CAM99_Y=czCKWSmgFqfZZJ+68FkqA_z+X65BF2K18CHbgF2XGtw@mail.gmail.com>
	<5D156F9A-BEC7-433E-BF74-A6654C417DBD@freeswitch.org>
	<CAM99_Y=SVtSeQZVXGoCZ9fZMDfVthH3JJtVTTbN3OXxpgcQBOA@mail.gmail.com>
	<CAGVOjVU4VcoO=62kpTBxyopOO1z1Vq85kHENd9jL+p5fQtd7vw@mail.gmail.com>
	<CAM99_Ymvih91h0aT06PVodK72THPXyR9upJJTM_EqyfZrak2cg@mail.gmail.com>
	<C4862E63-64CD-4DA9-A30A-99FFAF7F6D06@freeswitch.org>
	<CAM99_YnOrpxKgbwM14mMjuN0e1TwOf4AMcedte2puWPRg-qR3g@mail.gmail.com>
	<CAMtXxQm5nnJ-wn-zZzpN4td3zqN+pL6fYg4AALsCuMMBoqwP=w@mail.gmail.com>
	<CAM99_YnhdHeE2KrKKN7Zoy=ECd1ACtXBRuEvLiQbAG+LKFrXjg@mail.gmail.com>
Message-ID: <CAMtXxQnBfCf7zkHhEe+sNtYuLA6N0zayBp8uAjsbmDj68Nai5Q@mail.gmail.com>

I'll check it out.

Thanks!

On Mon, Jun 13, 2016 at 2:56 AM, Deepika Yadav <deepikay at iiitd.ac.in> wrote:

> You can follow the mod_python Wiki :
>
> https://wiki.freeswitch.org/wiki/Mod_python
>
> Inside the handler()
>
> session.hangup()
> new_api_obj = API()
> command_string = "originate
> {sip_auth_username=1002,sip_auth_password=XXXXXXX}sofia/external/1002 at XX.XX.XX.XX
> "+dialled_number
> new_api_obj.executeString(command_string)
>
> Here, it creates a new session after previous session is hungup completely
> and makes an outbound call
>
>
> I want play my own IVR menu when dialling to the dialled_number, to do
> some actions on DTMF inputs.
> So, I use the command :
>
> command_string = "sofia/gateway/MySIP/"+callerid+" 4449"
>
> Here, it does not create  a new session and report as "USER_BUSY", since
> it did not hungup in case before creating a new channel
>
>
>
>
>
> On Mon, Jun 13, 2016 at 10:33 AM, Joel Serrano <joel at gogii.net> wrote:
>
>> Hi Deepika,
>>
>> Would you mind sharing how you did it? I'm kinda curious. Do you
>> subscribe to events through python and accordingly or do you run a python
>> script from the dialplan?
>>
>> I'd love to see your python script as I normally use LUA and I want to
>> give python a try.
>>
>> Thanks,
>> Joel.
>>
>>
>>
>> On Sun, Jun 12, 2016 at 8:02 PM, Deepika Yadav <deepikay at iiitd.ac.in>
>> wrote:
>>
>>> Thanks Ken,
>>>
>>> I have figured it out, I used a Python module that reject the call by
>>> hangup() and then created a new session for outgoing call.
>>> On 13-Jun-2016 12:39 am, "Ken Rice" <krice at freeswitch.org> wrote:
>>>
>>>> Ok let me try this again. Once the incoming call is hung up, the call
>>>> is dead and anything processing on the channel (except the CDR) is done. No
>>>> further commands in the dial-plan will execute.
>>>>
>>>>
>>>> If you need pro help with this contact consulting at freeswitch.org and
>>>> one of us pros can knock this out pretty fast for you
>>>>
>>>> Sent from my iPhone
>>>>
>>>> On Jun 12, 2016, at 12:01 PM, Deepika Yadav <deepikay at iiitd.ac.in>
>>>> wrote:
>>>>
>>>> Even if I use Hangup command dialplan won't execute the statements
>>>> after hangup.
>>>>
>>>> My basic requirement is when Freeswitch receives a call on an
>>>> extension, it should reject it and then callback.
>>>>
>>>> On Sun, Jun 12, 2016 at 9:00 PM, Aqs Younas <aqsyounas at gmail.com>
>>>> wrote:
>>>>
>>>>> You can hangup the call with reason.
>>>>> https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_hangup
>>>>> On Jun 12, 2016 7:36 PM, "Deepika Yadav" <deepikay at iiitd.ac.in> wrote:
>>>>>
>>>>>> Even in ESL, first I have to reject the call and start a new session.
>>>>>>
>>>>>> What is the command to reject a call in mod_python, I tried
>>>>>> session.respond() that did not work.
>>>>>>
>>>>>>
>>>>>> Python script :
>>>>>>
>>>>>> def handler(session, args):
>>>>>>
>>>>>>     namer = session.getVariable('caller_id_number')
>>>>>>     freeswitch.consoleLog('info','Called by '+namer)
>>>>>>     session.respond()
>>>>>>     s = freeswitch.Session();
>>>>>>     s.originate(session,"
>>>>>> {sip_auth_username=1002,sip_auth_password=XXXXXXX}
>>>>>> sofia/external/1002 at XX.XX.XX.XX <sofia/external/1002 at xx.xx.xx.xx>
>>>>>>  9868789909")
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Sun, Jun 12, 2016 at 12:12 AM, Ken Rice <krice at freeswitch.org>
>>>>>> wrote:
>>>>>>
>>>>>>> Look at your logs you'll notice that originate is not a dialplan
>>>>>>> command and after you hang up on the incoming call the channel is dead. The
>>>>>>> correct way to do this is with a script (probably ESl) that sees the
>>>>>>> incoming call hangs it up and triggers the originate command
>>>>>>>
>>>>>>> Sent from my iPhone
>>>>>>>
>>>>>>> On Jun 11, 2016, at 1:26 PM, Deepika Yadav <deepikay at iiitd.ac.in>
>>>>>>> wrote:
>>>>>>>
>>>>>>> sorry for the incomplete message, please ignore the previous message.
>>>>>>>
>>>>>>> I am creating the dialplan in this way:
>>>>>>>
>>>>>>> <include>
>>>>>>>   <extension name="public_did">
>>>>>>>     <condition field="destination_number" expression="^(88888888)$">
>>>>>>>
>>>>>>> <action application="set" data="callerID = $${caller_id_number}"/>
>>>>>>> <action application="log" data="INFO src number is $${callerID}"/>
>>>>>>>         <action application="respond" data="503"/>
>>>>>>> <action application="originate"
>>>>>>> data="{sip_auth_username=1003,sip_auth_password=*******}sofia/external/
>>>>>>> 1003@ <1003 at 52.38.67.78>X.X.X.X $callerID "/>
>>>>>>>     </condition>
>>>>>>>   </extension>
>>>>>>> </include>
>>>>>>>
>>>>>>>
>>>>>>> <extension name="show_prepare">
>>>>>>> <condition field="destination_number"
>>>>>>> expression="^(?:\+91|0|)(\d{10})$">
>>>>>>>
>>>>>>> <action application="bridge" data="sofia/gateway/MySIP/91$1"/>
>>>>>>>       </condition>
>>>>>>> </extension>
>>>>>>>
>>>>>>> After rejecting the call, new channel for outgoing call does not
>>>>>>> start , also variable "callerID" is set to empty not capturing the
>>>>>>> source number
>>>>>>>
>>>>>>> Regards,
>>>>>>> Deepika
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Sat, Jun 11, 2016 at 11:50 PM, Deepika Yadav <
>>>>>>> deepikay at iiitd.ac.in> wrote:
>>>>>>>
>>>>>>>> Hi,
>>>>>>>>
>>>>>>>> I want to reject an incoming call from a mobile number in the
>>>>>>>> public context and callback to it
>>>>>>>>
>>>>>>>> how should the dialplan look like:
>>>>>>>>
>>>>>>>>
>>>>>>>> <include>
>>>>>>>>   <extension name="public_did">
>>>>>>>>     <condition field="destination_number"
>>>>>>>> expression="^(88888888)$">
>>>>>>>> <action application="set" data="domain_name=$${domain}"/>
>>>>>>>> <action application="log" data="INFO domain is $${domain}"/>
>>>>>>>> <action application="set" data="callerID = $${caller_id_number}"/>
>>>>>>>> <action application="log" data="INFO src number is $${callerID}"/>
>>>>>>>> <action application="respond" data="503"/>
>>>>>>>> <action application="bridge"
>>>>>>>> data="{sip_auth_username=1003,sip_auth_password=R at D
>>>>>>>> ~o$121$@IIITD}sofia/external/1003 at 52.38.67.78 9769658019"/>
>>>>>>>> <!--<action application="respond" data="reponse_info" data="407"/>
>>>>>>>> -->
>>>>>>>> <!--<action application="transfer" data="5000 XML default"/> -->
>>>>>>>>
>>>>>>>>     </condition>
>>>>>>>>   </extension>
>>>>>>>> </include>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Regards,
>>>>>>>> Deepika
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://
>>>>>>> lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://
>>>> lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 --------------
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From vittico at gmail.com  Mon Jun 13 21:31:30 2016
From: vittico at gmail.com (Victor Medina)
Date: Mon, 13 Jun 2016 13:31:30 -0400
Subject: [Freeswitch-users] ICE issue
In-Reply-To: <CAL+i1s9RFUry9XEzGAp81jJcPp-VdfNX=BtZ-=8UufdyhRum2Q@mail.gmail.com>
References: <CAL+i1s-_Sg8iN+ygKS0yVhM=_cdvvtC29at3L=5hHzcWyE3f8Q@mail.gmail.com>
	<CAL+i1s8at0j19ZjUnYVve_NDLaHLh5Gzbze=KYaw5v4j7_VYTA@mail.gmail.com>
	<CAEA-bfHwFWopNNvGpYj-8oOLe9EBJ-UA4LQDX73w5fkZYx-6Gg@mail.gmail.com>
	<CAEA-bfGHCH51VZbODgTeEX5kGd1n+ZRociW9d76gwGck1UXqHw@mail.gmail.com>
	<CAL+i1s9V6ZYJYb64VNdwjgB0z83He1a7Rk8uaqnESL3_SVuiUg@mail.gmail.com>
	<CAEA-bfG3qoeFjUzD0uFn4BfS=-JE_Z6iSJFAPHoY_9xBkkuZjQ@mail.gmail.com>
	<CAL+i1s9RFUry9XEzGAp81jJcPp-VdfNX=BtZ-=8UufdyhRum2Q@mail.gmail.com>
Message-ID: <CAEA-bfGSAWYihwMhd2hRp9faWAiygjPNVqkRjzVqHiqcJrT5pQ@mail.gmail.com>

In vars.XML. whats in your inbound and outbound c?dec list?
El 12/6/2016 6:26, "Bastien LE REST" <ba.lerest at gmail.com> escribi?:

> Hello Victor,
>
> Yes, I configured FS to only use iLBC codec with mod_iLBC. iOS works fine
> using this codec but I still have the same problem with android.
>
> Here's the trace: https://pastebin.freeswitch.org/view/fcf3b334
>
> Thank you for your help, it's much appreciated.
>
> Kind regards,
> Bastien.
>
> On 11 June 2016 at 21:22, Victor Medina <vittico at gmail.com> wrote:
>
>> Are they enabled con FS side??
>> El 11/6/2016 5:39, "Bastien LE REST" <ba.lerest at gmail.com> escribi?:
>>
>>> Hello Victor,
>>>
>>> Thank you for your reply. I tried to add both iLBC and Speex as a
>>> priority codec in the codecs list but unfortunately that didn't fix the
>>> problem.
>>>
>>> Is it normal that FS doesn't find a candidate for RTCP?
>>>
>>> Kind regards,
>>> Bastien.
>>>
>>> On 10 June 2016 at 19:33, Victor Medina <vittico at gmail.com> wrote:
>>>
>>>> More...
>>>>
>>>> On iOS you seem to be sending:
>>>>
>>>>
>>>>    1.    a=rtpmap:98 speex/16000
>>>>    2.    a=rtpmap:97 speex/8000
>>>>    3.    a=rtpmap:99 speex/32000
>>>>    4.    a=rtpmap:104 iLBC/8000
>>>>    5.    a=fmtp:104 mode=30
>>>>    6.    a=rtpmap:3 GSM/8000
>>>>    7.    a=rtpmap:0 PCMU/8000
>>>>    8.    a=rtpmap:8 PCMA/8000
>>>>    9.    a=rtpmap:9 G722/8000
>>>>
>>>>
>>>> While on Android I can only see:
>>>>
>>>>
>>>>    1.    a=rtpmap:98 speex/16000
>>>>    2.    a=rtpmap:97 speex/8000
>>>>    3.    a=rtpmap:99 speex/32000
>>>>    4.    a=rtpmap:104 iLBC/8000
>>>>
>>>>
>>>> FS is correctly complaining about it:
>>>>
>>>> 2016-06-10 12:48:33.533730 [ERR] mod_sofia.c:2346 CODEC NEGOTIATION
>>>> ERROR.  SDP:
>>>>
>>>>
>>>> Sin mas a que hacer referencia,
>>>>
>>>> Victor Medina
>>>>
>>>> On Fri, Jun 10, 2016 at 2:26 PM, Victor Medina <vittico at gmail.com>
>>>> wrote:
>>>>
>>>>> There is seems to be a codec issue.
>>>>> Are you sure codecs are right on the Android side?
>>>>>
>>>>>
>>>>> Sin mas a que hacer referencia,
>>>>>
>>>>> Victor Medina
>>>>>
>>>>> On Fri, Jun 10, 2016 at 1:28 PM, Bastien LE REST <ba.lerest at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> Bonjour,
>>>>>>
>>>>>> I thought the issue was solved since the calls work just fine now
>>>>>> with PJSIP on iOS, however the calls don't work with PJSIP on Android and I
>>>>>> don't understand why, the PJSIP configuration is exactly the same on both
>>>>>> devices.
>>>>>>
>>>>>> FS finds a candidate for RTP but it doesn't find a candidate for
>>>>>> RTCP. Please note that I only have 3 candidate.
>>>>>>
>>>>>> FS trace: https://pastebin.freeswitch.org/view/8d4df983
>>>>>>
>>>>>> Could you please have a look at the trace, and let me know what am I
>>>>>> doing wrong?
>>>>>>
>>>>>> Thank you very much in advance for your help in this matter.
>>>>>>
>>>>>> Kind regards,
>>>>>>
>>>>>> Bastien.
>>>>>>
>>>>>> On 4 June 2016 at 13:44, Bastien LE REST <ba.lerest at gmail.com> wrote:
>>>>>>
>>>>>>> Hi,
>>>>>>>
>>>>>>> When I enable ICE using PJSIP, FreeSwitch answers "488 Not
>>>>>>> Acceptable Here". When I disable ICE, I don't have any problems, the calls
>>>>>>> work just fine. I tried to use another SIP client with ICE and the call
>>>>>>> works fine. So I think the problem has to do with PJSIP.
>>>>>>>
>>>>>>> I'm working on the migration from an Asterisk architecture to a
>>>>>>> FreeSwitch architecture. I have to deal with PJSIP and ICE support because
>>>>>>> my client is a mobile application who is already published on the stores.
>>>>>>>
>>>>>>> PJSIP trace : https://pastebin.freeswitch.org/view/ddeaae4b
>>>>>>> FreeSwitch trace: https://pastebin.freeswitch.org/view/d380e01d
>>>>>>>
>>>>>>> Could you please have a look at the trace and tell me what am I
>>>>>>> doing wrong? Thank you in advance.
>>>>>>>
>>>>>>> Kind regards,
>>>>>>> Bastien.
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ba.lerest at gmail.com  Mon Jun 13 22:26:24 2016
From: ba.lerest at gmail.com (Bastien LE REST)
Date: Mon, 13 Jun 2016 19:26:24 +0100
Subject: [Freeswitch-users] ICE issue
In-Reply-To: <CAEA-bfGSAWYihwMhd2hRp9faWAiygjPNVqkRjzVqHiqcJrT5pQ@mail.gmail.com>
References: <CAL+i1s-_Sg8iN+ygKS0yVhM=_cdvvtC29at3L=5hHzcWyE3f8Q@mail.gmail.com>
	<CAL+i1s8at0j19ZjUnYVve_NDLaHLh5Gzbze=KYaw5v4j7_VYTA@mail.gmail.com>
	<CAEA-bfHwFWopNNvGpYj-8oOLe9EBJ-UA4LQDX73w5fkZYx-6Gg@mail.gmail.com>
	<CAEA-bfGHCH51VZbODgTeEX5kGd1n+ZRociW9d76gwGck1UXqHw@mail.gmail.com>
	<CAL+i1s9V6ZYJYb64VNdwjgB0z83He1a7Rk8uaqnESL3_SVuiUg@mail.gmail.com>
	<CAEA-bfG3qoeFjUzD0uFn4BfS=-JE_Z6iSJFAPHoY_9xBkkuZjQ@mail.gmail.com>
	<CAL+i1s9RFUry9XEzGAp81jJcPp-VdfNX=BtZ-=8UufdyhRum2Q@mail.gmail.com>
	<CAEA-bfGSAWYihwMhd2hRp9faWAiygjPNVqkRjzVqHiqcJrT5pQ@mail.gmail.com>
Message-ID: <CAL+i1s95Uj3Md-FEvQ2joVjm+npHR3DKeVBn5rM_Dz+oJzv01w@mail.gmail.com>

Hi Victor,

I think the problem is the transcoding. Maybe the iOS call works fine
because there is the codec G722 available on both legs.

Please find my FS configuration here. The configuration is from scratch.
All other parameters value are set by default.

FreeSWITCH Version 1.6.8+git~20160505T153832Z~99de0ad502~64bit (git 99de0ad
2016-05-05 15:38:32Z 64bit)

*vars.xml*

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,speex at 8000h
> @20i,speex at 16000h@20i,speex at 32000h@20i"/>

<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722"/>


*internal.xml*

<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>

<param name="inbound-late-negotiation" value="false"/>
> <param name="inbound-zrtp-passthru" value="false"/>
> <param name="disable-transcoding" value="false"/>
> <param name="inherit_codec" value="false"/>
> <param name="bypass_media" value="false"/>
> <param name="proxy_media" value="false"/>


*dialplan*

<action application="bridge"
> data="{media_mix_inbound_outbound_codecs=true,absolute_codec_string='G722'}sofia/gateway/
> sip.nl.didlogic.net/${destination_number}"/>


*show codec *

codec,G.722,mod_spandsp
> codec,Speex,CORE_SPEEX_MODULE
> codec,iLBC,mod_ilbc


*sofia status profile internal*

Name             internal
> Domain Name       N/A
> Auto-NAT         false
> DBName           sofia_reg_internal
> Pres Hosts       188.166.40.202,188.166.40.202
> Dialplan         XML
> Context           public
> Challenge Realm   auto_from
> RTP-IP           188.166.40.202
> SIP-IP           188.166.40.202
> URL               sip:mod_sofia at 188.166.40.202:5060
> BIND-URL         sip:mod_sofia at 188.166.40.202:5060;transport=udp,tcp
> WS-BIND-URL     sip:mod_sofia at 188.166.40.202:5066;transport=ws
> WSS-BIND-URL     sips:mod_sofia at 188.166.40.202:7443;transport=wss
> HOLD-MUSIC       local_stream://moh
> OUTBOUND-PROXY   N/A
> CODECS IN       OPUS,speex at 8000h@20i,speex at 16000h@20i,speex at 32000h@20i
> CODECS OUT                 G722
> TEL-EVENT                     101
> DTMF-MODE                   rfc2833
> CNG                                13
> SESSION-TO                  0
> MAX-DIALOG                 0
> NOMEDIA                       false
> LATE-NEG                      false
> PROXY-MEDIA               false
> ZRTP-PASSTHRU          false
> AGGRESSIVENAT         false
> CALLS-IN                       1
> FAILED-CALLS-IN         1
> CALLS-OUT                   0
> FAILED-CALLS-OUT     0
> REGISTRATIONS          1


Thank you for you help.

Kind regards,
Bastien.

On 13 June 2016 at 18:31, Victor Medina <vittico at gmail.com> wrote:

> In vars.XML. whats in your inbound and outbound c?dec list?
> El 12/6/2016 6:26, "Bastien LE REST" <ba.lerest at gmail.com> escribi?:
>
>> Hello Victor,
>>
>> Yes, I configured FS to only use iLBC codec with mod_iLBC. iOS works fine
>> using this codec but I still have the same problem with android.
>>
>> Here's the trace: https://pastebin.freeswitch.org/view/fcf3b334
>>
>> Thank you for your help, it's much appreciated.
>>
>> Kind regards,
>> Bastien.
>>
>> On 11 June 2016 at 21:22, Victor Medina <vittico at gmail.com> wrote:
>>
>>> Are they enabled con FS side??
>>> El 11/6/2016 5:39, "Bastien LE REST" <ba.lerest at gmail.com> escribi?:
>>>
>>>> Hello Victor,
>>>>
>>>> Thank you for your reply. I tried to add both iLBC and Speex as a
>>>> priority codec in the codecs list but unfortunately that didn't fix the
>>>> problem.
>>>>
>>>> Is it normal that FS doesn't find a candidate for RTCP?
>>>>
>>>> Kind regards,
>>>> Bastien.
>>>>
>>>> On 10 June 2016 at 19:33, Victor Medina <vittico at gmail.com> wrote:
>>>>
>>>>> More...
>>>>>
>>>>> On iOS you seem to be sending:
>>>>>
>>>>>
>>>>>    1.    a=rtpmap:98 speex/16000
>>>>>    2.    a=rtpmap:97 speex/8000
>>>>>    3.    a=rtpmap:99 speex/32000
>>>>>    4.    a=rtpmap:104 iLBC/8000
>>>>>    5.    a=fmtp:104 mode=30
>>>>>    6.    a=rtpmap:3 GSM/8000
>>>>>    7.    a=rtpmap:0 PCMU/8000
>>>>>    8.    a=rtpmap:8 PCMA/8000
>>>>>    9.    a=rtpmap:9 G722/8000
>>>>>
>>>>>
>>>>> While on Android I can only see:
>>>>>
>>>>>
>>>>>    1.    a=rtpmap:98 speex/16000
>>>>>    2.    a=rtpmap:97 speex/8000
>>>>>    3.    a=rtpmap:99 speex/32000
>>>>>    4.    a=rtpmap:104 iLBC/8000
>>>>>
>>>>>
>>>>> FS is correctly complaining about it:
>>>>>
>>>>> 2016-06-10 12:48:33.533730 [ERR] mod_sofia.c:2346 CODEC NEGOTIATION
>>>>> ERROR.  SDP:
>>>>>
>>>>>
>>>>> Sin mas a que hacer referencia,
>>>>>
>>>>> Victor Medina
>>>>>
>>>>> On Fri, Jun 10, 2016 at 2:26 PM, Victor Medina <vittico at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> There is seems to be a codec issue.
>>>>>> Are you sure codecs are right on the Android side?
>>>>>>
>>>>>>
>>>>>> Sin mas a que hacer referencia,
>>>>>>
>>>>>> Victor Medina
>>>>>>
>>>>>> On Fri, Jun 10, 2016 at 1:28 PM, Bastien LE REST <ba.lerest at gmail.com
>>>>>> > wrote:
>>>>>>
>>>>>>> Bonjour,
>>>>>>>
>>>>>>> I thought the issue was solved since the calls work just fine now
>>>>>>> with PJSIP on iOS, however the calls don't work with PJSIP on Android and I
>>>>>>> don't understand why, the PJSIP configuration is exactly the same on both
>>>>>>> devices.
>>>>>>>
>>>>>>> FS finds a candidate for RTP but it doesn't find a candidate for
>>>>>>> RTCP. Please note that I only have 3 candidate.
>>>>>>>
>>>>>>> FS trace: https://pastebin.freeswitch.org/view/8d4df983
>>>>>>>
>>>>>>> Could you please have a look at the trace, and let me know what am I
>>>>>>> doing wrong?
>>>>>>>
>>>>>>> Thank you very much in advance for your help in this matter.
>>>>>>>
>>>>>>> Kind regards,
>>>>>>>
>>>>>>> Bastien.
>>>>>>>
>>>>>>> On 4 June 2016 at 13:44, Bastien LE REST <ba.lerest at gmail.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> Hi,
>>>>>>>>
>>>>>>>> When I enable ICE using PJSIP, FreeSwitch answers "488 Not
>>>>>>>> Acceptable Here". When I disable ICE, I don't have any problems, the calls
>>>>>>>> work just fine. I tried to use another SIP client with ICE and the call
>>>>>>>> works fine. So I think the problem has to do with PJSIP.
>>>>>>>>
>>>>>>>> I'm working on the migration from an Asterisk architecture to a
>>>>>>>> FreeSwitch architecture. I have to deal with PJSIP and ICE support because
>>>>>>>> my client is a mobile application who is already published on the stores.
>>>>>>>>
>>>>>>>> PJSIP trace : https://pastebin.freeswitch.org/view/ddeaae4b
>>>>>>>> FreeSwitch trace: https://pastebin.freeswitch.org/view/d380e01d
>>>>>>>>
>>>>>>>> Could you please have a look at the trace and tell me what am I
>>>>>>>> doing wrong? Thank you in advance.
>>>>>>>>
>>>>>>>> Kind regards,
>>>>>>>> Bastien.
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jmoran at secureachsystems.com  Mon Jun 13 22:59:35 2016
From: jmoran at secureachsystems.com (Jason Moran)
Date: Mon, 13 Jun 2016 14:59:35 -0400
Subject: [Freeswitch-users] mod_v8 - still no File object?
Message-ID: <05db01d1c5a5$bc945b10$35bd1130$@secureachsystems.com>

I've been running old FS w/Spidermonkey, finally upgraded and testing in
mod_v8.

My javascript returned this:

 

Exception: ReferenceError: File is not defined (near: "var fd = new
File(myPath + myFileName")

 

I read about the lack of XML and File back in 2015, but I assumed it has
been included since spidermonkey no longer is an option. No? Other options?

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From krice at freeswitch.org  Mon Jun 13 23:05:37 2016
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 13 Jun 2016 14:05:37 -0500
Subject: [Freeswitch-users] mod_v8 - still no File object?
In-Reply-To: <05db01d1c5a5$bc945b10$35bd1130$@secureachsystems.com>
References: <05db01d1c5a5$bc945b10$35bd1130$@secureachsystems.com>
Message-ID: <028301d1c5a6$92938740$b7ba95c0$@freeswitch.org>

Features are added based on need of the user bases. This is something that
has only recently come up. If this is something you may be interested in
sponsoring email consulting at freeswitch.org
<mailto:consulting at freeswitch.org>  or if you have the required skills to
patch the C for this a pull request for this sort of feature would almost
always be accepted after a standard code review.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason
Moran
Sent: Monday, June 13, 2016 2:00 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] mod_v8 - still no File object?

 

I've been running old FS w/Spidermonkey, finally upgraded and testing in
mod_v8.

My javascript returned this:

 

Exception: ReferenceError: File is not defined (near: "var fd = new
File(myPath + myFileName")

 

I read about the lack of XML and File back in 2015, but I assumed it has
been included since spidermonkey no longer is an option. No? Other options?

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From mike at jerris.com  Mon Jun 13 23:37:24 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 13 Jun 2016 15:37:24 -0400
Subject: [Freeswitch-users] mod_v8 - still no File object?
In-Reply-To: <05db01d1c5a5$bc945b10$35bd1130$@secureachsystems.com>
References: <05db01d1c5a5$bc945b10$35bd1130$@secureachsystems.com>
Message-ID: <D09EA261-E6D0-492E-9CD4-4A04A688CA44@jerris.com>

Double check the code in mod_v8/src dir, there are File and FileIO and XML objects... see what they are missing.. the docs are:

https://freeswitch.org/confluence/display/FREESWITCH/JavaScript+API+Reference <https://freeswitch.org/confluence/display/FREESWITCH/JavaScript+API+Reference>

not sure what all is missing in the docs, I suspect those are incomplete.

> On Jun 13, 2016, at 2:59 PM, Jason Moran <jmoran at secureachsystems.com> wrote:
> 
> I?ve been running old FS w/Spidermonkey, finally upgraded and testing in mod_v8.
> My javascript returned this:
>  
> Exception: ReferenceError: File is not defined (near: "var fd = new File(myPath + myFileName")
>  
> I read about the lack of XML and File back in 2015, but I assumed it has been included since spidermonkey no longer is an option. No? Other options?

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From joel at gogii.net  Tue Jun 14 02:00:46 2016
From: joel at gogii.net (Joel Serrano)
Date: Mon, 13 Jun 2016 15:00:46 -0700
Subject: [Freeswitch-users] bridge and originate / multiple dests and
	failover
Message-ID: <CAMtXxQnyS-JAbr7vkEf9gvrY44vBsLGyGew4kFiMHogArFtmFg@mail.gmail.com>

Hi,

Is it possible to bridge using both multiple destinations and failover?

For example:

Is the order failover > group:

{ignore_early_media=true}sofia/profile/number1 at trunk1
|{ignore_early_media=true}sofia/profile/number1 at trunk2
,{ignore_early_media=true}sofia/profile/number2 at trunk1
|{ignore_early_media=true}sofia/profile/number1 at trunk2

Or maybe group > failover:

{ignore_early_media=true}sofia/profile/number1 at trunk1
,{ignore_early_media=true}sofia/profile/number2 at trunk1
|{ignore_early_media=true}sofia/profile/number1 at trunk2,
{ignore_early_media=true}sofia/profile/number2 at trunk2

??

Is this also equally valid when run from originate ?


Thanks,
Joel.
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From mike at jerris.com  Tue Jun 14 02:16:40 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 13 Jun 2016 18:16:40 -0400
Subject: [Freeswitch-users] bridge and originate / multiple dests and
	failover
In-Reply-To: <CAMtXxQnyS-JAbr7vkEf9gvrY44vBsLGyGew4kFiMHogArFtmFg@mail.gmail.com>
References: <CAMtXxQnyS-JAbr7vkEf9gvrY44vBsLGyGew4kFiMHogArFtmFg@mail.gmail.com>
Message-ID: <167C3771-0AF4-40EE-B2CB-B4683C79B8FF@jerris.com>

you can use a combo of ands and ors .. yes  you only have one set of { }.. { } get assigned to the a leg

> On Jun 13, 2016, at 6:00 PM, Joel Serrano <joel at gogii.net> wrote:
> 
> Hi, 
> 
> Is it possible to bridge using both multiple destinations and failover?
> 
> For example:
> 
> Is the order failover > group:
> 
> {ignore_early_media=true}sofia/profile/number1 at trunk1|{ignore_early_media=true}sofia/profile/number1 at trunk2,{ignore_early_media=true}sofia/profile/number2 at trunk1|{ignore_early_media=true}sofia/profile/number1 at trunk2
> 
> Or maybe group > failover:
> 
> {ignore_early_media=true}sofia/profile/number1 at trunk1,{ignore_early_media=true}sofia/profile/number2 at trunk1|{ignore_early_media=true}sofia/profile/number1 at trunk2,
> {ignore_early_media=true}sofia/profile/number2 at trunk2
> 
> ?? 
> 
> Is this also equally valid when run from originate ?
> 
> 
> Thanks, 
> Joel.
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From joel at gogii.net  Tue Jun 14 02:27:02 2016
From: joel at gogii.net (Joel Serrano)
Date: Mon, 13 Jun 2016 15:27:02 -0700
Subject: [Freeswitch-users] bridge and originate / multiple dests and
	failover
In-Reply-To: <167C3771-0AF4-40EE-B2CB-B4683C79B8FF@jerris.com>
References: <CAMtXxQnyS-JAbr7vkEf9gvrY44vBsLGyGew4kFiMHogArFtmFg@mail.gmail.com>
	<167C3771-0AF4-40EE-B2CB-B4683C79B8FF@jerris.com>
Message-ID: <CAMtXxQmfnRCFO2Fce4ijmj6BLof7mzD+JrSNyAtQtMaHhmurpg@mail.gmail.com>

Hi Michael,

So the correct format would be

{all_aleg_vars}sofia/profile/number at trunk AND / OR
sofia/profile/number at trunk AND / OR sofia/profile/number at trunk

etc...

Only having one set of {} for the a-leg and they would be applied to all of
the calls right? (simultaneous and failovers)

Thanks,
Joel.


On Mon, Jun 13, 2016 at 3:16 PM, Michael Jerris <mike at jerris.com> wrote:

> you can use a combo of ands and ors .. yes  you only have one set of { }..
> { } get assigned to the a leg
>
> > On Jun 13, 2016, at 6:00 PM, Joel Serrano <joel at gogii.net> wrote:
> >
> > Hi,
> >
> > Is it possible to bridge using both multiple destinations and failover?
> >
> > For example:
> >
> > Is the order failover > group:
> >
> > {ignore_early_media=true}sofia/profile/number1 at trunk1
> |{ignore_early_media=true}sofia/profile/number1 at trunk2
> ,{ignore_early_media=true}sofia/profile/number2 at trunk1
> |{ignore_early_media=true}sofia/profile/number1 at trunk2
> >
> > Or maybe group > failover:
> >
> > {ignore_early_media=true}sofia/profile/number1 at trunk1
> ,{ignore_early_media=true}sofia/profile/number2 at trunk1
> |{ignore_early_media=true}sofia/profile/number1 at trunk2,
> > {ignore_early_media=true}sofia/profile/number2 at trunk2
> >
> > ??
> >
> > Is this also equally valid when run from originate ?
> >
> >
> > Thanks,
> > Joel.
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Tue Jun 14 02:52:35 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 13 Jun 2016 18:52:35 -0400
Subject: [Freeswitch-users] bridge and originate / multiple dests and
	failover
In-Reply-To: <CAMtXxQmfnRCFO2Fce4ijmj6BLof7mzD+JrSNyAtQtMaHhmurpg@mail.gmail.com>
References: <CAMtXxQnyS-JAbr7vkEf9gvrY44vBsLGyGew4kFiMHogArFtmFg@mail.gmail.com>
	<167C3771-0AF4-40EE-B2CB-B4683C79B8FF@jerris.com>
	<CAMtXxQmfnRCFO2Fce4ijmj6BLof7mzD+JrSNyAtQtMaHhmurpg@mail.gmail.com>
Message-ID: <CCA9952B-FF76-43AC-B75B-27DD7B442359@jerris.com>

no the , and | is right.. i was referring to what they did.  The { } are applied only to the a leg.. [ ] vars are applied to the individual b legs 


> On Jun 13, 2016, at 6:27 PM, Joel Serrano <joel at gogii.net> wrote:
> 
> Hi Michael, 
> 
> So the correct format would be 
> 
> {all_aleg_vars}sofia/profile/number at trunk AND / OR sofia/profile/number at trunk AND / OR sofia/profile/number at trunk 
> 
> etc...
> 
> Only having one set of {} for the a-leg and they would be applied to all of the calls right? (simultaneous and failovers)
> 
> Thanks,
> Joel.
> 
> 
> On Mon, Jun 13, 2016 at 3:16 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> you can use a combo of ands and ors .. yes  you only have one set of { }.. { } get assigned to the a leg
> 
> > On Jun 13, 2016, at 6:00 PM, Joel Serrano <joel at gogii.net <mailto:joel at gogii.net>> wrote:
> >
> > Hi,
> >
> > Is it possible to bridge using both multiple destinations and failover?
> >
> > For example:
> >
> > Is the order failover > group:
> >
> > {ignore_early_media=true}sofia/profile/number1 at trunk1|{ignore_early_media=true}sofia/profile/number1 at trunk2,{ignore_early_media=true}sofia/profile/number2 at trunk1|{ignore_early_media=true}sofia/profile/number1 at trunk2
> >
> > Or maybe group > failover:
> >
> > {ignore_early_media=true}sofia/profile/number1 at trunk1,{ignore_early_media=true}sofia/profile/number2 at trunk1|{ignore_early_media=true}sofia/profile/number1 at trunk2,
> > {ignore_early_media=true}sofia/profile/number2 at trunk2
> >
> > ??
> >
> > Is this also equally valid when run from originate ?
> >
> >
> > Thanks,
> > Joel.
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> > http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org <http://www.freeswitch.org/>
> > http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> > http://www.cluecon.com <http://www.cluecon.com/>
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> > http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From joel at gogii.net  Tue Jun 14 03:17:31 2016
From: joel at gogii.net (Joel Serrano)
Date: Mon, 13 Jun 2016 16:17:31 -0700
Subject: [Freeswitch-users] bridge and originate / multiple dests and
	failover
In-Reply-To: <CCA9952B-FF76-43AC-B75B-27DD7B442359@jerris.com>
References: <CAMtXxQnyS-JAbr7vkEf9gvrY44vBsLGyGew4kFiMHogArFtmFg@mail.gmail.com>
	<167C3771-0AF4-40EE-B2CB-B4683C79B8FF@jerris.com>
	<CAMtXxQmfnRCFO2Fce4ijmj6BLof7mzD+JrSNyAtQtMaHhmurpg@mail.gmail.com>
	<CCA9952B-FF76-43AC-B75B-27DD7B442359@jerris.com>
Message-ID: <CAMtXxQ=R-11bVz75kL5jcq-NMEOCm19W8YRD7beqp27MHDAWYQ@mail.gmail.com>

Ok! Gotya this time.

If it's in a bridge, ok cause you already have an active a-leg pending to
be bridged to a b-leg, however, in case its an originate instead of bridge,
all of them would be A-legs... can you have different {} for each one or
the first one would get applied to all ?

Thanks again!
Joel.



On Mon, Jun 13, 2016 at 3:52 PM, Michael Jerris <mike at jerris.com> wrote:

> no the , and | is right.. i was referring to what they did.  The { } are
> applied only to the a leg.. [ ] vars are applied to the individual b legs
>
>
> On Jun 13, 2016, at 6:27 PM, Joel Serrano <joel at gogii.net> wrote:
>
> Hi Michael,
>
> So the correct format would be
>
> {all_aleg_vars}sofia/profile/number at trunk AND / OR
> sofia/profile/number at trunk AND / OR sofia/profile/number at trunk
>
> etc...
>
> Only having one set of {} for the a-leg and they would be applied to all
> of the calls right? (simultaneous and failovers)
>
> Thanks,
> Joel.
>
>
> On Mon, Jun 13, 2016 at 3:16 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> you can use a combo of ands and ors .. yes  you only have one set of {
>> }.. { } get assigned to the a leg
>>
>> > On Jun 13, 2016, at 6:00 PM, Joel Serrano <joel at gogii.net> wrote:
>> >
>> > Hi,
>> >
>> > Is it possible to bridge using both multiple destinations and failover?
>> >
>> > For example:
>> >
>> > Is the order failover > group:
>> >
>> > {ignore_early_media=true}sofia/profile/number1 at trunk1
>> |{ignore_early_media=true}sofia/profile/number1 at trunk2
>> ,{ignore_early_media=true}sofia/profile/number2 at trunk1
>> |{ignore_early_media=true}sofia/profile/number1 at trunk2
>> >
>> > Or maybe group > failover:
>> >
>> > {ignore_early_media=true}sofia/profile/number1 at trunk1
>> ,{ignore_early_media=true}sofia/profile/number2 at trunk1
>> |{ignore_early_media=true}sofia/profile/number1 at trunk2,
>> > {ignore_early_media=true}sofia/profile/number2 at trunk2
>> >
>> > ??
>> >
>> > Is this also equally valid when run from originate ?
>> >
>> >
>> > Thanks,
>> > Joel.
>> >
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dujinfang at gmail.com  Tue Jun 14 05:34:19 2016
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 14 Jun 2016 09:34:19 +0800
Subject: [Freeswitch-users] mod_v8 - still no File object?
In-Reply-To: <D09EA261-E6D0-492E-9CD4-4A04A688CA44@jerris.com>
References: <05db01d1c5a5$bc945b10$35bd1130$@secureachsystems.com>
	<D09EA261-E6D0-492E-9CD4-4A04A688CA44@jerris.com>
Message-ID: <66EEEF8B400E4F53B2B72172EA331659@gmail.com>

@Jason I think I have a local patch not sure if it solves your issue. but if you would like to test and make pull request create a jira I can attach the patch.  


--  
Seven Du
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Tuesday, June 14, 2016 at 3:37 AM, Michael Jerris wrote:

> Double check the code in mod_v8/src dir, there are File and FileIO and XML objects... see what they are missing.. the docs are:
>  
> https://freeswitch.org/confluence/display/FREESWITCH/JavaScript+API+Reference
>  
> not sure what all is missing in the docs, I suspect those are incomplete.
>  
> > On Jun 13, 2016, at 2:59 PM, Jason Moran <jmoran at secureachsystems.com (mailto:jmoran at secureachsystems.com)> wrote:
> > I?ve been running old FS w/Spidermonkey, finally upgraded and testing in mod_v8.
> > My javascript returned this:
> >   
> > Exception: ReferenceError: File is not defined (near: "var fd = new File(myPath + myFileName")
> >   
> > I read about the lack of XML and File back in 2015, but I assumed it has been included since spidermonkey no longer is an option. No? Other options?
> >  
> >  
> >  
>  
>  
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:  
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>  
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org)
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
>  


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From lte at lte-net.de  Tue Jun 14 13:14:53 2016
From: lte at lte-net.de (Fred Schulz)
Date: Tue, 14 Jun 2016 11:14:53 +0200
Subject: [Freeswitch-users] libSRTP version issue
Message-ID: <4D1A9B26-0F27-4E14-9D32-E5990B917212@lte-net.de>

Hi all,

 

as I can see freeSWITCH is using the cisco srtp reference implemenation. The version in FS 1.4.26 is 1.4.3 which is a little bit old and vulnerable by DoS Attacks according to this:

 

https://cve.mitre.org/cgi-bin/cvename.cgi?name=CVE-2015-6360

 

I just wanted to ask if the freeSWITCH will use the Version 1.5.4 or 2.0.0-pre which can be found here:

 

https://github.com/cisco/libsrtp/

 

 

 

I just tried to use the ?new? one while compiling but there are some declaration and invalig operator errors while making.

 

?

In file included from /usr/local/src/freeswitch/libs/srtp/crypto/include/datatypes.h:50:0,

                 from src/switch_rtp.c:53:

/usr/local/src/freeswitch/libs/srtp/crypto/include/integers.h:102:16: error: conflicting types for 'uint64_t'

In file included from /usr/lib/gcc/x86_64-linux-gnu/4.7/include/stdint.h:3:0,

                 from /usr/include/inttypes.h:28,

                 from ./src/include/switch_platform.h:123,

                 from ./src/include/switch.h:112,

                 from src/switch_rtp.c:39:

/usr/include/stdint.h:56:27: note: previous declaration of 'uint64_t' was here

In file included from src/switch_rtp.c:55:0:

/usr/local/src/freeswitch/libs/srtp/include/srtp_priv.h:82:3: error: type of bit-field 'cc' is a GCC extension [-Werror=edantic]

/usr/local/src/freeswitch/libs/srtp/include/srtp_priv.h:83:3: error: type of bit-field 'x' is a GCC extension [-Werror=edantic]

/usr/local/src/freeswitch/libs/srtp/include/srtp_priv.h:84:3: error: type of bit-field 'p' is a GCC extension [-Werror=edantic]

/usr/local/src/freeswitch/libs/srtp/include/srtp_priv.h:85:3: error: type of bit-field 'version' is a GCC extension [-Werror=edantic]

/usr/local/src/freeswitch/libs/srtp/include/srtp_priv.h:86:3: error: type of bit-field 'pt' is a GCC extension [-Werror=edantic]

/usr/local/src/freeswitch/libs/srtp/include/srtp_priv.h:87:3: error: type of bit-field 'm' is a GCC extension [-Werror=edantic]

/usr/local/src/freeswitch/libs/srtp/include/srtp_priv.h:125:3: error: type of bit-field 'rc' is a GCC extension [-Werror=edantic]

/usr/local/src/freeswitch/libs/srtp/include/srtp_priv.h:126:3: error: type of bit-field 'p' is a GCC extension [-Werror=edantic]

/usr/local/src/freeswitch/libs/srtp/include/srtp_priv.h:127:3: error: type of bit-field 'version' is a GCC extension [-Werror=edantic]

/usr/local/src/freeswitch/libs/srtp/include/srtp_priv.h:128:3: error: type of bit-field 'pt' is a GCC extension [-Werror=edantic]

src/switch_rtp.c: In function 'rtcp_generate_sender_info':

src/switch_rtp.c:1864:16: error: invalid operands to binary << (have 'double' and 'int')

cc1: all warnings being treated as errors

make[1]: *** [src/libfreeswitch_la-switch_rtp.lo] Fehler 1

make[1]: Leaving directory `/usr/local/src/freeswitch'

make: *** [all] Fehler 2

?

 

Is there something planned?

 

Thanks you

Fred Schulz

blackned GmbH

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From alex at digitalmail.com  Tue Jun 14 15:14:17 2016
From: alex at digitalmail.com (Alex Lake)
Date: Tue, 14 Jun 2016 11:14:17 +0000
Subject: [Freeswitch-users] Please don't be cross.... (Question relating to
	FS1.2!)
Message-ID: <4f868a1a-25c6-f217-8a99-f1ec6a92311b@digitalmail.com>

...but how do I make FS1.2 use Lua5.2?

Maybe some of you with longer memories will know ;-)

Alex

(And, yes, I'm currently battling with my hosting provider to go for a 
Debian 8 build with FS1.6!)


From mike at jerris.com  Tue Jun 14 17:33:59 2016
From: mike at jerris.com (Michael Jerris)
Date: Tue, 14 Jun 2016 09:33:59 -0400
Subject: [Freeswitch-users] libSRTP version issue
In-Reply-To: <4D1A9B26-0F27-4E14-9D32-E5990B917212@lte-net.de>
References: <4D1A9B26-0F27-4E14-9D32-E5990B917212@lte-net.de>
Message-ID: <CAEwTsAis66BKnFh8fSU04=oL3aaSi_c-yR=7snDad=t7xoJTKQ@mail.gmail.com>

please file a jira for this, I'll need to review this

On Tuesday, June 14, 2016, Fred Schulz <lte at lte-net.de> wrote:

> Hi all,
>
>
>
> as I can see freeSWITCH is using the cisco srtp reference implemenation.
> The version in FS 1.4.26 is 1.4.3 which is a little bit old and vulnerable
> by DoS Attacks according to this:
>
>
>
> https://cve.mitre.org/cgi-bin/cvename.cgi?name=CVE-2015-6360
>
>
>
> I just wanted to ask if the freeSWITCH will use the Version 1.5.4 or
> 2.0.0-pre which can be found here:
>
>
>
> https://github.com/cisco/libsrtp/
>
>
>
>
>
>
>
> I just tried to use the ?new? one while compiling but there are some
> declaration and invalig operator errors while making.
>
>
>
> ?
>
> In file included from
> /usr/local/src/freeswitch/libs/srtp/crypto/include/datatypes.h:50:0,
>
>                  from src/switch_rtp.c:53:
>
> /usr/local/src/freeswitch/libs/srtp/crypto/include/integers.h:102:16:
> error: conflicting types for 'uint64_t'
>
> In file included from
> /usr/lib/gcc/x86_64-linux-gnu/4.7/include/stdint.h:3:0,
>
>                  from /usr/include/inttypes.h:28,
>
>                  from ./src/include/switch_platform.h:123,
>
>                  from ./src/include/switch.h:112,
>
>                  from src/switch_rtp.c:39:
>
> /usr/include/stdint.h:56:27: note: previous declaration of 'uint64_t' was
> here
>
> In file included from src/switch_rtp.c:55:0:
>
> /usr/local/src/freeswitch/libs/srtp/include/srtp_priv.h:82:3: error: type
> of bit-field 'cc' is a GCC extension [-Werror=edantic]
>
> /usr/local/src/freeswitch/libs/srtp/include/srtp_priv.h:83:3: error: type
> of bit-field 'x' is a GCC extension [-Werror=edantic]
>
> /usr/local/src/freeswitch/libs/srtp/include/srtp_priv.h:84:3: error: type
> of bit-field 'p' is a GCC extension [-Werror=edantic]
>
> /usr/local/src/freeswitch/libs/srtp/include/srtp_priv.h:85:3: error: type
> of bit-field 'version' is a GCC extension [-Werror=edantic]
>
> /usr/local/src/freeswitch/libs/srtp/include/srtp_priv.h:86:3: error: type
> of bit-field 'pt' is a GCC extension [-Werror=edantic]
>
> /usr/local/src/freeswitch/libs/srtp/include/srtp_priv.h:87:3: error: type
> of bit-field 'm' is a GCC extension [-Werror=edantic]
>
> /usr/local/src/freeswitch/libs/srtp/include/srtp_priv.h:125:3: error: type
> of bit-field 'rc' is a GCC extension [-Werror=edantic]
>
> /usr/local/src/freeswitch/libs/srtp/include/srtp_priv.h:126:3: error: type
> of bit-field 'p' is a GCC extension [-Werror=edantic]
>
> /usr/local/src/freeswitch/libs/srtp/include/srtp_priv.h:127:3: error: type
> of bit-field 'version' is a GCC extension [-Werror=edantic]
>
> /usr/local/src/freeswitch/libs/srtp/include/srtp_priv.h:128:3: error: type
> of bit-field 'pt' is a GCC extension [-Werror=edantic]
>
> src/switch_rtp.c: In function 'rtcp_generate_sender_info':
>
> src/switch_rtp.c:1864:16: error: invalid operands to binary << (have
> 'double' and 'int')
>
> cc1: all warnings being treated as errors
>
> make[1]: *** [src/libfreeswitch_la-switch_rtp.lo] Fehler 1
>
> make[1]: Leaving directory `/usr/local/src/freeswitch'
>
> make: *** [all] Fehler 2
>
> ?
>
>
>
> Is there something planned?
>
>
>
> Thanks you
>
> Fred Schulz
>
> blackned GmbH
>
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From mike at jerris.com  Tue Jun 14 17:36:41 2016
From: mike at jerris.com (Michael Jerris)
Date: Tue, 14 Jun 2016 09:36:41 -0400
Subject: [Freeswitch-users] Please don't be cross.... (Question relating
 to FS1.2!)
In-Reply-To: <4f868a1a-25c6-f217-8a99-f1ec6a92311b@digitalmail.com>
References: <4f868a1a-25c6-f217-8a99-f1ec6a92311b@digitalmail.com>
Message-ID: <CAEwTsAhP_SgV8chJ8XbZudSLPbuwvQsvpS1+i13qd4d-23qiSg@mail.gmail.com>

it will require a bunch of code changes for sure.  I wouldn't bother
trying.  If you want to use newer lua you should be using FreeSWITCH 1.6.
FreeSWITCH 1.2 has significant know security issues and continuing to use
it puts you at significant risk, we encourage people to use much more
recent releases.

On Tuesday, June 14, 2016, Alex Lake <alex at digitalmail.com> wrote:

> ...but how do I make FS1.2 use Lua5.2?
>
> Maybe some of you with longer memories will know ;-)
>
> Alex
>
> (And, yes, I'm currently battling with my hosting provider to go for a
> Debian 8 build with FS1.6!)
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <javascript:;>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From alex at digitalmail.com  Tue Jun 14 17:46:54 2016
From: alex at digitalmail.com (Alex Lake)
Date: Tue, 14 Jun 2016 13:46:54 +0000
Subject: [Freeswitch-users] Please don't be cross.... (Question relating
	to FS1.2!)
In-Reply-To: <CAEwTsAhP_SgV8chJ8XbZudSLPbuwvQsvpS1+i13qd4d-23qiSg@mail.gmail.com>
References: <4f868a1a-25c6-f217-8a99-f1ec6a92311b@digitalmail.com>
	<CAEwTsAhP_SgV8chJ8XbZudSLPbuwvQsvpS1+i13qd4d-23qiSg@mail.gmail.com>
Message-ID: <6d57a841-23b7-69ec-d43c-0153903bb8b4@digitalmail.com>

Thanks - that's very useful. FS 1.6 currently installing as we speak....

On 14/06/2016 14:36, Michael Jerris wrote:
> it will require a bunch of code changes for sure.  I wouldn't bother 
> trying.  If you want to use newer lua you should be using FreeSWITCH 
> 1.6.  FreeSWITCH 1.2 has significant know security issues and 
> continuing to use it puts you at significant risk, we encourage people 
> to use much more recent releases.
>
> On Tuesday, June 14, 2016, Alex Lake <alex at digitalmail.com 
> <mailto:alex at digitalmail.com>> wrote:
>
>     ...but how do I make FS1.2 use Lua5.2?
>
>     Maybe some of you with longer memories will know ;-)
>
>     Alex
>
>     (And, yes, I'm currently battling with my hosting provider to go for a
>     Debian 8 build with FS1.6!)
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <javascript:;>
>     http://www.freeswitchsolutions.com
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org <javascript:;>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



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From stefan.mititelu92 at gmail.com  Tue Jun 14 16:48:29 2016
From: stefan.mititelu92 at gmail.com (Mititelu Stefan)
Date: Tue, 14 Jun 2016 15:48:29 +0300
Subject: [Freeswitch-users] Freeswitch "playback"
Message-ID: <CAL82Oa0EkKj1QT5mKdrL234xGLTu=SgTSWEfuJE7w0=SFVEWWA@mail.gmail.com>

Hi,

I'm trying to do [1] I get [2] even if wget on that link works, on that
machine. It works with local .wav though.

Do I need to do something else in addition? Do you have some links that
work for you?

Thanks,
Stefan


[1] <action application="playback" data="
http://bbcmedia.ic.llnwd.net/stream/bbcmedia_radio2_mf_p"/>
[2] 2016-06-14 12:11:58.430013 [ERR] mod_httapi.c:2793 File at url [
http://bbcmedia.ic.llnwd.net/stream/bbcmedia_radio2_mf_p] is unreachable!
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From daveh at beachdognet.com  Tue Jun 14 18:44:22 2016
From: daveh at beachdognet.com (Dave Horton)
Date: Tue, 14 Jun 2016 10:44:22 -0400
Subject: [Freeswitch-users] possible to access packet count on a channel?
Message-ID: <F00F8C82-9B31-49AD-822B-4AF502D8B666@beachdognet.com>

Is there any way to access the number of RTP packets received on a channel?  This could be accomplished by accessing the RTCP information / stats that are maintained.  Is anything like that possible?

Dave

From lists at kavun.ch  Tue Jun 14 18:46:50 2016
From: lists at kavun.ch (Emrah)
Date: Tue, 14 Jun 2016 16:46:50 +0200
Subject: [Freeswitch-users] distributed presence (again)
In-Reply-To: <28106D4F-2A5F-4EFD-993A-83207A1D650B@beachdognet.com>
References: <5B0BFA80-9BA4-46BD-AD70-5170F48D1CE8@beachdognet.com>
	<79B2BC2C-4B26-43F5-A46C-9EC961D0F704@kavun.ch>
	<28106D4F-2A5F-4EFD-993A-83207A1D650B@beachdognet.com>
Message-ID: <DC07E3BA-FDA7-434A-BDAF-9EC5655CEC1D@kavun.ch>

I think your idea makes perfect sense. That?d be a great addition indeed.

> On Jun 13, 2016, at 2:09 PM, Dave Horton <daveh at beachdognet.com> wrote:
> 
> Wouldn?t it make sense for their to be a concept of a ?presence realm? which is different than a hostname (i,e, a grouping of multiple hosts)?
> On Jun 13, 2016, at 7:34 AM, Emrah <lists at kavun.ch> wrote:
> 
> Thanks for this post Dave. Likewise I?m looking for a solution. Looks like DB support is all there already. Let me know how I can help.
> If we are missing something and distributed presence is already functional, some info would be appreciated.
> I cannot share the same hostname because each server must keep its own registrations in order for my distributed setup to work.
> 
> Emrah
>> On Jun 10, 2016, at 4:21 AM, Dave Horton <daveh at beachdognet.com> wrote:
>> 
>> I?ve seen a few recent posts on distributed presence work, and I am running into the same need.
>> Like the other?s who have posted, I can?t use the same hostname across servers.
>> In my case, this is because in order to make call transfer scenarios work I need
>> to redirect calls sometimes to specific servers, and unique hostnames are required for that.
>> 
>> Is there another solution for this?  
>> 
>> If not, I?d be interested in working on a solution, as long as there was a view towards
>> incorporating it in the mainstream code at some point (i.e., I wouldn't want to have to 
>> keep patching releases going forward).
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: 
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From benjamin.cropley at gmail.com  Tue Jun 14 18:53:36 2016
From: benjamin.cropley at gmail.com (Benjamin Cropley)
Date: Tue, 14 Jun 2016 15:53:36 +0100
Subject: [Freeswitch-users] Freeswitch "playback"
In-Reply-To: <CAL82Oa0EkKj1QT5mKdrL234xGLTu=SgTSWEfuJE7w0=SFVEWWA@mail.gmail.com>
References: <CAL82Oa0EkKj1QT5mKdrL234xGLTu=SgTSWEfuJE7w0=SFVEWWA@mail.gmail.com>
Message-ID: <CAF7jiz+yLntmc_dXE1BJNUQOT2Bo+ES5gca2qKN+HcPDJeeuRw@mail.gmail.com>

Did you do wget as the same user FS runs under?

Either way, it may be a good idea to do a packet capture (google tcpdump)
and check you're 100% making a request for that file.

On Tue, Jun 14, 2016 at 1:48 PM, Mititelu Stefan <
stefan.mititelu92 at gmail.com> wrote:

> Hi,
>
> I'm trying to do [1] I get [2] even if wget on that link works, on that
> machine. It works with local .wav though.
>
> Do I need to do something else in addition? Do you have some links that
> work for you?
>
> Thanks,
> Stefan
>
>
> [1] <action application="playback" data="
> http://bbcmedia.ic.llnwd.net/stream/bbcmedia_radio2_mf_p"/>
> [2] 2016-06-14 12:11:58.430013 [ERR] mod_httapi.c:2793 File at url [
> http://bbcmedia.ic.llnwd.net/stream/bbcmedia_radio2_mf_p] is unreachable!
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lists at kavun.ch  Tue Jun 14 18:57:11 2016
From: lists at kavun.ch (Emrah)
Date: Tue, 14 Jun 2016 16:57:11 +0200
Subject: [Freeswitch-users] Cannot query mod_db with a variable
Message-ID: <A049A330-6BB9-4CDE-9100-23E6DD3CD043@kavun.ch>

Hi list,
When I do this,
<action application=?set? data=?db_data=${db(select/foo/${dialed_number})}? inline=?true?/>
I get this:
SET sofia/internal/user at domain [db_data]=[!err!]

Likewise when I try to evaluate the content of a key in a condition. It works only if I don?t use a variable. I must be screwing up the syntax somewhere?

Any way I could make this work?
Emrah
FreeSWITCH Version 1.7.0-829

From tanim05 at gmail.com  Tue Jun 14 19:14:05 2016
From: tanim05 at gmail.com (Tanim)
Date: Tue, 14 Jun 2016 21:14:05 +0600
Subject: [Freeswitch-users] How to playback background call process
Message-ID: <000701d1c64f$65ff6f10$31fe4d30$@gmail.com>

Hi,

 

I want to playback file background call process. So that in playing IVR
time, call will process to send to gateways. Currently my dialplan like
this:

<action application="playback" data="/IVR/FILE" />

<action application="bridge" data="sofia/gateway/gw1/123456" />

 

In this way before call processing IVR is playing then start to process the
call. But I want both in same time.

Can you please help me, how can I do it.

 

Thanks in advance.

Tanim

 

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From steveayre at gmail.com  Tue Jun 14 19:15:55 2016
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 14 Jun 2016 16:15:55 +0100
Subject: [Freeswitch-users] possible to access packet count on a channel?
In-Reply-To: <F00F8C82-9B31-49AD-822B-4AF502D8B666@beachdognet.com>
References: <F00F8C82-9B31-49AD-822B-4AF502D8B666@beachdognet.com>
Message-ID: <CAFiqYunYfxAfrOWPgcvtDc2RydHd0_ZxiFJPRjy_cNBi+CDddA@mail.gmail.com>

It's in the XML CDR at the end of the call as the rtp_audio_in_packet_count
variable.

During the call you can probably get it by doing 'uuid_set_media_stats
<uuid>' followed by 'uuid_getvar <uuid> rtp_audio_in_packet_count'

On 14 June 2016 at 15:44, Dave Horton <daveh at beachdognet.com> wrote:

> Is there any way to access the number of RTP packets received on a
> channel?  This could be accomplished by accessing the RTCP information /
> stats that are maintained.  Is anything like that possible?
>
> Dave
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Tue Jun 14 20:17:12 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 14 Jun 2016 11:17:12 -0500
Subject: [Freeswitch-users] Freeswitch "playback"
In-Reply-To: <CAF7jiz+yLntmc_dXE1BJNUQOT2Bo+ES5gca2qKN+HcPDJeeuRw@mail.gmail.com>
References: <CAL82Oa0EkKj1QT5mKdrL234xGLTu=SgTSWEfuJE7w0=SFVEWWA@mail.gmail.com>
	<CAF7jiz+yLntmc_dXE1BJNUQOT2Bo+ES5gca2qKN+HcPDJeeuRw@mail.gmail.com>
Message-ID: <CAKbxfG-YobH8c4J-fbDqT1cEB2H2dU8rLp6aMM=FmW8QrvBzhQ@mail.gmail.com>

If this is a stream, mod_httapi is the wrong mod, you would need mod_shout.


On Tue, Jun 14, 2016 at 9:53 AM, Benjamin Cropley <
benjamin.cropley at gmail.com> wrote:

> Did you do wget as the same user FS runs under?
>
> Either way, it may be a good idea to do a packet capture (google tcpdump)
> and check you're 100% making a request for that file.
>
> On Tue, Jun 14, 2016 at 1:48 PM, Mititelu Stefan <
> stefan.mititelu92 at gmail.com> wrote:
>
>> Hi,
>>
>> I'm trying to do [1] I get [2] even if wget on that link works, on that
>> machine. It works with local .wav though.
>>
>> Do I need to do something else in addition? Do you have some links that
>> work for you?
>>
>> Thanks,
>> Stefan
>>
>>
>> [1] <action application="playback" data="
>> http://bbcmedia.ic.llnwd.net/stream/bbcmedia_radio2_mf_p"/>
>> [2] 2016-06-14 12:11:58.430013 [ERR] mod_httapi.c:2793 File at url [
>> http://bbcmedia.ic.llnwd.net/stream/bbcmedia_radio2_mf_p] is unreachable!
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From david.villasmil.work at gmail.com  Tue Jun 14 22:22:16 2016
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 14 Jun 2016 20:22:16 +0200
Subject: [Freeswitch-users] FS as a SRS
In-Reply-To: <CA+d2mV-P13CYNOUedJ9qKOU1RU4heL13Pzg8=agYyZk5v0phDg@mail.gmail.com>
References: <CAFGRPVqfkVxgfSrUWQXmNWEcSm5X+7_68u=x4qt-pgNXo-PbLw@mail.gmail.com>
	<CA+d2mV-5pG-XCy1w=sCNk5gMduy6row1YYM+7zS7dkfomqyq2w@mail.gmail.com>
	<CAFGRPVoGxQvn=PSK1Miq5QkpjrZWROoGmXSzf+XMh6titnct3A@mail.gmail.com>
	<CA+d2mV-P13CYNOUedJ9qKOU1RU4heL13Pzg8=agYyZk5v0phDg@mail.gmail.com>
Message-ID: <CAFGRPVoUWDw3+74MrQ2quZfzF-UGBYtmk4HQT8vdvRR2WfzJsg@mail.gmail.com>

Hello,

I see, well thanks, anyway. I can probably user pjsip :)

Regards,

David
?

On Sun, Jun 12, 2016 at 9:47 AM, Miguel Oyarzo <miguelaustro at gmail.com>
wrote:

> Hi David,
> I got you, but FS can't accept multiple m=audio lines, which makes it
> incompatible for SIPREC.
> I don't think there is plans to modify the SIP diver in freeswitch, in
> order to
> support this protocol. Enabling additional RTP engines for a single SDP
> session is not trivial.
>
> Regards,
>
>
>
> On Wed, Jun 8, 2016 at 10:03 PM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Hello Miguel,
>>
>> Thanks for your reply, but I think maybe I didn't explain myself
>> correctly, let me try again:
>>
>> A recording server receives INVITEs including multiple "m=" which will be
>> used as rtp source so the server can record an ongoing conversation
>> happening somewhere else. As per RFC4566 an SDP may have as many m= as it
>> wants (https://tools.ietf.org/html/rfc4566#page-22).
>>
>> The idea is to use SIPREC the following way:
>>
>> The voice server will choose the call to record and send a 3rd leg to the
>> recording server, which will receive the INVITE ans answer with 200 OK and
>> start recording the rtps.
>>
>> Regards,
>>
>> David
>>
>> On Wed, Jun 8, 2016 at 12:40 PM, Miguel Oyarzo <miguelaustro at gmail.com>
>> wrote:
>>
>>>
>>>
>>> Nighter FS nor Asterisk will process two same-type RTP streams in the
>>> same SDP. The 2nd RTP streams will be declined and answered with a port 0
>>> (200OK), as suggested by rfc3264.
>>>
>>> Regards,
>>>
>>> On Wed, Jun 8, 2016 at 2:37 AM, David Villasmil <
>>> david.villasmil.work at gmail.com> wrote:
>>>
>>>> Hello,
>>>> Is it possible to use FS as a recording server?
>>>>
>>>> We will be receiving INVITEs with a siprec tag and sdp with the
>>>> participants. To which FS should answer with 2 rtp ports on which to
>>>> receive the audio. We would be receiving 2 rtp streams that we need to
>>>> record to a file.
>>>>
>>>> Thanks!
>>>>
>>>> David
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> "Do not use a cannon to kill a mosquito" (Confucius)
>>>
>>> Miguel Oyarzo
>>> VoIP Engineer && DevOps
>>> Linux User: # 483188 - counter.li.org
>>> http://au.linkedin.com/in/mikeaustralia
>>> Melbourne, Australia
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> "Do not use a cannon to kill a mosquito" (Confucius)
>
> Miguel Oyarzo
> VoIP Engineer && DevOps
> Linux User: # 483188 - counter.li.org
> http://au.linkedin.com/in/mikeaustralia
> Melbourne, Australia
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From aqsyounas at gmail.com  Tue Jun 14 22:44:49 2016
From: aqsyounas at gmail.com (Aqs Younas)
Date: Tue, 14 Jun 2016 23:44:49 +0500
Subject: [Freeswitch-users] [ERR] helloworld.js:8 Exception:
 ReferenceError: File is not defined
In-Reply-To: <CAPEdYTidc-Fj4mrcTm-BKyjPoO7nLGbrE_tR=2k0jA17e1ZVNA@mail.gmail.com>
References: <CAGVOjVXuXhdEbjrggx6aXmkADZ4Y1vXL0vEMaNLrwVfTdewdpg@mail.gmail.com>
	<CAPEdYTidc-Fj4mrcTm-BKyjPoO7nLGbrE_tR=2k0jA17e1ZVNA@mail.gmail.com>
Message-ID: <CAGVOjVUYWZNCdns_nP8GLCs0G3cs2bK=WCHV-51GruGOCLU+Cg@mail.gmail.com>

Thanks for your reply.

But the file exists on the specified path, I can open the file using FileIO
but when i try to use File i get this error.

Any pointer?

On 10 June 2016 at 17:41, Peter Olsson <peter at olssononline.se> wrote:

> As referenced here, "File" is not available in mod_v8.
>
> https://freeswitch.org/confluence/display/FREESWITCH/JavaScript
>
> /Peter
>
> 2016-06-09 20:28 GMT+02:00 Aqs Younas <aqsyounas at gmail.com>:
>
>> Greetings list.
>>
>> I am getting below error when i am trying open a file in mod_v8.
>>
>> 2016-06-02 01:40:44.908007 [ERR] helloworld.js:8 Exception:
>> ReferenceError: File is not defined (near: "        fd = new
>> File(baseFolder + "/" + menuId + ".json");")
>>
>> Any pointer is much appreciated
>>
>> Thanks.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From joel at gogii.net  Tue Jun 14 23:58:49 2016
From: joel at gogii.net (Joel Serrano)
Date: Tue, 14 Jun 2016 12:58:49 -0700
Subject: [Freeswitch-users] FS 1.6.9 deb packages
Message-ID: <CAMtXxQmgLiLEPozgPDA0__Y_wOgAZ2kvk0PZX9xYPeDFH0rbdA@mail.gmail.com>

Hi,

I've seen that v1.6.9 is out, but I don't see the packages in the repo, is
it just because they are still being build or are they not going to be
available for this version?

Thanks,
Joel.
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From cody.rosenbloom at gmail.com  Wed Jun 15 02:47:35 2016
From: cody.rosenbloom at gmail.com (Cody Rosenbloom)
Date: Tue, 14 Jun 2016 18:47:35 -0400
Subject: [Freeswitch-users] Testing for call hangup in dialplan xml
In-Reply-To: <6839235B-178F-43C9-8D17-5D740D2DB17D@freeswitch.org>
References: <CAPrxTnqAneapO9Zpfx73sMGOGdV9mFrtZaEjgmV_rjzhG-gn0g@mail.gmail.com>
	<6839235B-178F-43C9-8D17-5D740D2DB17D@freeswitch.org>
Message-ID: <CAPrxTnpcrmThxFsPasOvjdo94gupfKvg6R4GQXF7=HJU19gWOQ@mail.gmail.com>

Thanks Ken, I've only just got a chance to look at this again, can you
provide more details about this hook you mentioned in terms of name and
location(if I sound like a total noob, its because I am, sorry!)? I'm
looking at voicemail.conf.xml under autoload_configs but haven't spotted
anything.

Thanks
Cody

On 8 June 2016 at 19:01, Ken Rice <krice at freeswitch.org> wrote:

> This is why we have things like CDRs that can trigger post call processing
> via web call or via file drop. If you are using mod_voicemail there is a
> post voicemail being left hook already exposed in the configs.
>
> Doing post call processing on the channel greatly increases the chances of
> hung channels resulting in system resources being held for dead calls
>
> Sent from my iPhone
>
> On Jun 8, 2016, at 5:54 PM, Cody Rosenbloom <cody.rosenbloom at gmail.com>
> wrote:
>
> Hi,
>
> Is there a way to check if call has been hung up in dialplan xml. I want
> to run lua script to process voicemail after caller has hungup the call. I
> tried checking if state is equal to CS_HANGUP but it does not work. Current
> dialplan xml is at http://pastebin.com/5PMKSrdh
>
> Thanks
> Cody
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From shishko69 at gmail.com  Wed Jun 15 09:39:09 2016
From: shishko69 at gmail.com (Denis Papes)
Date: Wed, 15 Jun 2016 05:39:09 +0000
Subject: [Freeswitch-users] Testing for call hangup in dialplan xml
In-Reply-To: <CAPrxTnpcrmThxFsPasOvjdo94gupfKvg6R4GQXF7=HJU19gWOQ@mail.gmail.com>
References: <CAPrxTnqAneapO9Zpfx73sMGOGdV9mFrtZaEjgmV_rjzhG-gn0g@mail.gmail.com>
	<6839235B-178F-43C9-8D17-5D740D2DB17D@freeswitch.org>
	<CAPrxTnpcrmThxFsPasOvjdo94gupfKvg6R4GQXF7=HJU19gWOQ@mail.gmail.com>
Message-ID: <CAHT_yt-e_OC=2WL0R40JYESkgRkg5PVYVMuAibAGRUF_HAZLXw@mail.gmail.com>

There is already post on this mailing list titled "Run a LUA script when
voicemail received" by Steven Schoch on May 27th solving same problem.
Check it out.

On Tue, Jun 14, 2016 at 11:49 PM Cody Rosenbloom <cody.rosenbloom at gmail.com>
wrote:

> Thanks Ken, I've only just got a chance to look at this again, can you
> provide more details about this hook you mentioned in terms of name and
> location(if I sound like a total noob, its because I am, sorry!)? I'm
> looking at voicemail.conf.xml under autoload_configs but haven't spotted
> anything.
>
> Thanks
> Cody
>
> On 8 June 2016 at 19:01, Ken Rice <krice at freeswitch.org> wrote:
>
>> This is why we have things like CDRs that can trigger post call
>> processing via web call or via file drop. If you are using mod_voicemail
>> there is a post voicemail being left hook already exposed in the configs.
>>
>> Doing post call processing on the channel greatly increases the chances
>> of hung channels resulting in system resources being held for dead calls
>>
>> Sent from my iPhone
>>
>> On Jun 8, 2016, at 5:54 PM, Cody Rosenbloom <cody.rosenbloom at gmail.com>
>> wrote:
>>
>> Hi,
>>
>> Is there a way to check if call has been hung up in dialplan xml. I want
>> to run lua script to process voicemail after caller has hungup the call. I
>> tried checking if state is equal to CS_HANGUP but it does not work. Current
>> dialplan xml is at http://pastebin.com/5PMKSrdh
>>
>> Thanks
>> Cody
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From a.m.morozov at gmail.com  Wed Jun 15 16:51:13 2016
From: a.m.morozov at gmail.com (=?UTF-8?B?0JDQu9C10LrRgdC10Lkg0JzQvtGA0L7Qt9C+0LI=?=)
Date: Wed, 15 Jun 2016 15:51:13 +0300
Subject: [Freeswitch-users] How to playback background call process
In-Reply-To: <000701d1c64f$65ff6f10$31fe4d30$@gmail.com>
References: <000701d1c64f$65ff6f10$31fe4d30$@gmail.com>
Message-ID: <CACJ9VGNY-hU59+S2mFzGeLO6YhHg7c53bjite6JTZXRQe3Bd9Q@mail.gmail.com>

Hi

Try to use <action application="set" data="ringback=/IVR/FILE" />



2016-06-14 18:14 GMT+03:00 Tanim <tanim05 at gmail.com>:

> Hi,
>
>
>
> I want to playback file background call process. So that in playing IVR
> time, call will process to send to gateways. Currently my dialplan like
> this:
>
> <action application="playback" data="/IVR/FILE" />
>
> <action application="bridge" data="sofia/gateway/gw1/123456" />
>
>
>
> In this way before call processing IVR is playing then start to process
> the call. But I want both in same time.
>
> Can you please help me, how can I do it.
>
>
>
> Thanks in advance.
>
> Tanim
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
WBR
Alex
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From tanim05 at gmail.com  Wed Jun 15 18:51:54 2016
From: tanim05 at gmail.com (Tanim)
Date: Wed, 15 Jun 2016 20:51:54 +0600
Subject: [Freeswitch-users] How to playback background call process
In-Reply-To: <CACJ9VGNY-hU59+S2mFzGeLO6YhHg7c53bjite6JTZXRQe3Bd9Q@mail.gmail.com>
References: <000701d1c64f$65ff6f10$31fe4d30$@gmail.com>
	<CACJ9VGNY-hU59+S2mFzGeLO6YhHg7c53bjite6JTZXRQe3Bd9Q@mail.gmail.com>
Message-ID: <00b801d1c715$76ef7420$64ce5c60$@gmail.com>

Thanks for your reply. Its worked partially.

When FS get ringback from far end then its working fine. But if a call take more time to get RBT then FS gets crashed. Here is the log:

 

[DEBUG] switch_ivr_originate.c:1305 Play Ringback File [file_string://digits/youhave.wav!digits/9.wav]

[DEBUG] switch_rtp.c:1846 rtcp_stats_init: audio ssrc[37546101] base_seq[0]

[DEBUG] switch_rtp.c:6707 Correct audio ip/port confirmed.

freeswitch: src/switch_core_file.c:378: switch_core_file_read: Assertion `fh->file_interface != ((void *)0)' failed.

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of ??????? ???????
Sent: Wednesday, June 15, 2016 6:51 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How to playback background call process

 

Hi

Try to use <action application="set" data="ringback=/IVR/FILE" />

 

 

2016-06-14 18:14 GMT+03:00 Tanim <tanim05 at gmail.com>:

Hi,

 

I want to playback file background call process. So that in playing IVR time, call will process to send to gateways. Currently my dialplan like this:

<action application="playback" data="/IVR/FILE" />

<action application="bridge" data="sofia/gateway/gw1/123456" />

 

In this way before call processing IVR is playing then start to process the call. But I want both in same time.

Can you please help me, how can I do it.

 

Thanks in advance.

Tanim

 


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

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

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




-- 

WBR

Alex 

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From me at nevian.org  Wed Jun 15 19:55:23 2016
From: me at nevian.org (Serge S. Yuriev)
Date: Wed, 15 Jun 2016 18:55:23 +0300
Subject: [Freeswitch-users] mod_ooh323 questions
Message-ID: <57617A6B.9060903@nevian.org>

Hello,

Is anyone using this module 'in wild'?
After short tests I found two things:
- Module generates Alerting just after receiving request regardless of 
status of outgoing leg even if we have only sip:100 provisional response
- Not translating q931 code from B to A: for example sip:404 resulted in 
OO_REASON_LOCAL_CLEARED. Same for INCOMPATIBLE DESTINATION

Is this something I can configure or it's flaws of underlying lib (as of 
discussion back in 2013) and means module unusable?

-- 
Serge S. Yuriev
Lead VoIP engineer



From eduardo at ctidealer.com.br  Wed Jun 15 21:51:40 2016
From: eduardo at ctidealer.com.br (edrperes)
Date: Wed, 15 Jun 2016 10:51:40 -0700 (MST)
Subject: [Freeswitch-users] GsmOpen call control
In-Reply-To: <1465401044177-7596222.post@n2.nabble.com>
References: <1465396074152-7596221.post@n2.nabble.com>
	<1465401044177-7596222.post@n2.nabble.com>
Message-ID: <1466013100593-7596226.post@n2.nabble.com>

I opened a JIRA to this question because i think it is as bug

I tried many variantions...

I put a hub with a 3 ampere supplye and nothing changed, for a single modem



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/GsmOpen-call-control-tp7596221p7596226.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From gmaruzz at gmail.com  Wed Jun 15 22:09:01 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 15 Jun 2016 20:09:01 +0200
Subject: [Freeswitch-users] GsmOpen call control
In-Reply-To: <1466013100593-7596226.post@n2.nabble.com>
References: <1465396074152-7596221.post@n2.nabble.com>
	<1465401044177-7596222.post@n2.nabble.com>
	<1466013100593-7596226.post@n2.nabble.com>
Message-ID: <CALXCt0rJw19OuWKeANe474-rNf_iui6QuTA=RsBePc7f7jvQDQ@mail.gmail.com>

put it in debug mode, and check if the modem expose those events.

I would suspect the modem, as is initialized, do not.

So, maybe you may want to find how to initialize that modem to make it
expose those events...

It is not a bug (obviously if events are exposed we use it), it can be a
non supported (eg, non correctly initialized) modem.

On Wed, Jun 15, 2016 at 7:51 PM, edrperes <eduardo at ctidealer.com.br> wrote:

> I opened a JIRA to this question because i think it is as bug
>
> I tried many variantions...
>
> I put a hub with a 3 ampere supplye and nothing changed, for a single modem
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/GsmOpen-call-control-tp7596221p7596226.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From lists at kavun.ch  Wed Jun 15 22:21:01 2016
From: lists at kavun.ch (Emrah)
Date: Wed, 15 Jun 2016 20:21:01 +0200
Subject: [Freeswitch-users] Dynamically turn on and off Voicemail and custom
	manage BLF
Message-ID: <075D04F6-F326-4EB9-971A-E1C31C6BB211@kavun.ch>

Hi list,
I am trying to emulate the behavior of a traditional PBX where pressing a soft key on the phone will forward calls to Voicemail and turn on the lamp on that key for all phones in the group.
I am able to switch on and off the forwarding to Voicemail by using a DB entry that is updated when you call a certain extension, but I am struggling with the BLF.
What I am doing currently is to send an event to turn on the BLF, and  one to turn off the BLF, as follows:
BLF on:
sendevent PRESENCE_IN
proto: sip
from: 1001 at domain.com
login: 1001 at domain.com
event_type: presence
alt_event_type: dialog
Presence-Call-Direction: outbound
answer-state: confirmed

BLF Off:
sendevent PRESENCE_IN
proto: sip
from: 1001 at domain.com
login: 1001 at domain.com
event_type: presence
alt_event_type: dialog
Presence-Call-Direction: outbound
answer-state: terminated

On the phone, the soft key is programmed to monitor 1001 at domain.com. 

1001 at domain.com shows as unregistered until I run one of the events on it. However, it doesn?t matter which I run, I am not able to control the on / off behavior on the BLF. The light comes on, and stays on until it looks as if the extension was unregistered again.

What am I missing and how can I accomplish this?

Thanks a million,
Emrah
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From eduardo at ctidealer.com.br  Wed Jun 15 22:26:35 2016
From: eduardo at ctidealer.com.br (eduardo at ctidealer.com.br)
Date: Wed, 15 Jun 2016 15:26:35 -0300
Subject: [Freeswitch-users] GsmOpen call control
In-Reply-To: <CALXCt0rJw19OuWKeANe474-rNf_iui6QuTA=RsBePc7f7jvQDQ@mail.gmail.com>
References: <1465396074152-7596221.post@n2.nabble.com>
	<1465401044177-7596222.post@n2.nabble.com>
	<1466013100593-7596226.post@n2.nabble.com>
	<CALXCt0rJw19OuWKeANe474-rNf_iui6QuTA=RsBePc7f7jvQDQ@mail.gmail.com>
Message-ID: <1b4e7a32a02c354175edde1e44247d1a@ctidealer.com.br>

 

How can I put in debug mode? 

Em 15.06.2016 15:09, Giovanni
Maruzzelli escreveu: 

> put it in debug mode, and check if the modem
expose those events.
> 
> I would suspect the modem, as is initialized,
do not.
> 
> So, maybe you may want to find how to initialize that modem
to make it expose those events...
> 
> It is not a bug (obviously if
events are exposed we use it), it can be a non supported (eg, non
correctly initialized) modem. 
> 
> On Wed, Jun 15, 2016 at 7:51 PM,
edrperes <eduardo at ctidealer.com.br> wrote:
> 
>> I opened a JIRA to this
question because i think it is as bug
>> 
>> I tried many
variantions...
>> 
>> I put a hub with a 3 ampere supply and nothing
changed, for a single modem
>> 
>> --
>> View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/GsmOpen-call-control-tp7596221p7596226.html
[1]
>> 
>> Sent from the freeswitch-users mailing list archive at
Nabble.com.
>> 
>>
_________________________________________________________________________
>>
Professional FreeSWITCH Consulting Services:
>>
consulting at freeswitch.org
>> http://www.freeswitchsolutions.com [2]
>>

>> Official FreeSWITCH Sites
>> http://www.freeswitch.org [3]
>>
http://confluence.freeswitch.org [4]
>> http://www.cluecon.com [5]
>>

>> FreeSWITCH-users mailing list
>>
FreeSWITCH-users at lists.freeswitch.org
>>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [6]
>>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[7]
>> http://www.freeswitch.org [3]
> 
> -- 
> 
> Sincerely,
> 
>
Giovanni Maruzzelli
> Cell : +39-347-2665618 
> 
>
_________________________________________________________________________
>
Professional FreeSWITCH Consulting Services: 
>
consulting at freeswitch.org
> http://www.freeswitchsolutions.com [2]
> 
>
Official FreeSWITCH Sites
> http://www.freeswitch.org [3]
>
http://confluence.freeswitch.org [4]
> http://www.cluecon.com [5]
> 
>
FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [6]
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[7]
> http://www.freeswitch.org [3]

 

Links:
------
[1]
http://freeswitch-users.2379917.n2.nabble.com/GsmOpen-call-control-tp7596221p7596226.html
[2]
http://www.freeswitchsolutions.com
[3] http://www.freeswitch.org
[4]
http://confluence.freeswitch.org
[5] http://www.cluecon.com
[6]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[7]
http://lists.freeswitch.org/mailman/options/freeswitch-users
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From peter at olssononline.se  Wed Jun 15 22:41:00 2016
From: peter at olssononline.se (Peter Olsson)
Date: Wed, 15 Jun 2016 20:41:00 +0200
Subject: [Freeswitch-users] [ERR] helloworld.js:8 Exception:
 ReferenceError: File is not defined
In-Reply-To: <CAGVOjVUYWZNCdns_nP8GLCs0G3cs2bK=WCHV-51GruGOCLU+Cg@mail.gmail.com>
References: <CAGVOjVXuXhdEbjrggx6aXmkADZ4Y1vXL0vEMaNLrwVfTdewdpg@mail.gmail.com>
	<CAPEdYTidc-Fj4mrcTm-BKyjPoO7nLGbrE_tR=2k0jA17e1ZVNA@mail.gmail.com>
	<CAGVOjVUYWZNCdns_nP8GLCs0G3cs2bK=WCHV-51GruGOCLU+Cg@mail.gmail.com>
Message-ID: <CAPEdYTig9hs+LguY-8c57BwVj3xLfPN7NCDcVnuoPFfbm=fPSw@mail.gmail.com>

I think you missunderstood me. The js "File" object doesn't exist in
mod_v8, so it can't be used in the script.

/Peter

2016-06-14 20:44 GMT+02:00 Aqs Younas <aqsyounas at gmail.com>:

> Thanks for your reply.
>
> But the file exists on the specified path, I can open the file using
> FileIO but when i try to use File i get this error.
>
> Any pointer?
>
> On 10 June 2016 at 17:41, Peter Olsson <peter at olssononline.se> wrote:
>
>> As referenced here, "File" is not available in mod_v8.
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/JavaScript
>>
>> /Peter
>>
>> 2016-06-09 20:28 GMT+02:00 Aqs Younas <aqsyounas at gmail.com>:
>>
>>> Greetings list.
>>>
>>> I am getting below error when i am trying open a file in mod_v8.
>>>
>>> 2016-06-02 01:40:44.908007 [ERR] helloworld.js:8 Exception:
>>> ReferenceError: File is not defined (near: "        fd = new
>>> File(baseFolder + "/" + menuId + ".json");")
>>>
>>> Any pointer is much appreciated
>>>
>>> Thanks.
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From eduardo at ctidealer.com.br  Wed Jun 15 22:45:38 2016
From: eduardo at ctidealer.com.br (eduardo at ctidealer.com.br)
Date: Wed, 15 Jun 2016 15:45:38 -0300
Subject: [Freeswitch-users] GsmOpen call control
In-Reply-To: <1b4e7a32a02c354175edde1e44247d1a@ctidealer.com.br>
References: <1465396074152-7596221.post@n2.nabble.com>
	<1465401044177-7596222.post@n2.nabble.com>
	<1466013100593-7596226.post@n2.nabble.com>
	<CALXCt0rJw19OuWKeANe474-rNf_iui6QuTA=RsBePc7f7jvQDQ@mail.gmail.com>
	<1b4e7a32a02c354175edde1e44247d1a@ctidealer.com.br>
Message-ID: <60dd418aa8ac524b68e17b1cb912fe9a@ctidealer.com.br>

 

This way? 

freeswitch at machine> console loglevel 9

freeswitch at machine> fsctl loglevel 9 
freeswitch at machine> load
mod_gsmopen 

Em 15.06.2016 15:26, eduardo at ctidealer.com.br escreveu:


> How can I put in debug mode? 
> 
> Em 15.06.2016 15:09, Giovanni
Maruzzelli escreveu: 
> 
>> put it in debug mode, and check if the modem
expose those events.
>> 
>> I would suspect the modem, as is
initialized, do not.
>> 
>> So, maybe you may want to find how to
initialize that modem to make it expose those events...
>> 
>> It is not
a bug (obviously if events are exposed we use it), it can be a non
supported (eg, non correctly initialized) modem. 
>> 
>> On Wed, Jun 15,
2016 at 7:51 PM, edrperes <eduardo at ctidealer.com.br> wrote:
>> 
>>> I
opened a JIRA to this question because i think it is as bug
>>> 
>>> I
tried many variantions...
>>> 
>>> I put a hub with a 3 ampere supply
and nothing changed, for a single modem
>>> 
>>> --
>>> View this
message in context:
http://freeswitch-users.2379917.n2.nabble.com/GsmOpen-call-control-tp7596221p7596226.html
[1]
>>> 
>>> Sent from the freeswitch-users mailing list archive at
Nabble.com.
>>> 
>>>
_________________________________________________________________________
>>>
Professional FreeSWITCH Consulting Services:
>>>
consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com [2]
>>>

>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org [3]
>>>
http://confluence.freeswitch.org [4]
>>> http://www.cluecon.com [5]
>>>

>>> FreeSWITCH-users mailing list
>>>
FreeSWITCH-users at lists.freeswitch.org
>>>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [6]
>>>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[7]
>>> http://www.freeswitch.org [3]
>> 
>> -- 
>> 
>> Sincerely,
>>

>> Giovanni Maruzzelli
>> Cell : +39-347-2665618 
>> 
>>
_________________________________________________________________________
>>
Professional FreeSWITCH Consulting Services: 
>>
consulting at freeswitch.org
>> http://www.freeswitchsolutions.com [2]
>>

>> Official FreeSWITCH Sites
>> http://www.freeswitch.org [3]
>>
http://confluence.freeswitch.org [4]
>> http://www.cluecon.com [5]
>>

>> FreeSWITCH-users mailing list
>>
FreeSWITCH-users at lists.freeswitch.org
>>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [6]
>>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[7]
>> http://www.freeswitch.org [3]

 

Links:
------
[1]
http://freeswitch-users.2379917.n2.nabble.com/GsmOpen-call-control-tp7596221p7596226.html
[2]
http://www.freeswitchsolutions.com
[3] http://www.freeswitch.org
[4]
http://confluence.freeswitch.org
[5] http://www.cluecon.com
[6]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[7]
http://lists.freeswitch.org/mailman/options/freeswitch-users
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From eduardo at ctidealer.com.br  Wed Jun 15 23:23:10 2016
From: eduardo at ctidealer.com.br (eduardo at ctidealer.com.br)
Date: Wed, 15 Jun 2016 16:23:10 -0300
Subject: [Freeswitch-users] GsmOpen call control
In-Reply-To: <60dd418aa8ac524b68e17b1cb912fe9a@ctidealer.com.br>
References: <1465396074152-7596221.post@n2.nabble.com>
	<1465401044177-7596222.post@n2.nabble.com>
	<1466013100593-7596226.post@n2.nabble.com>
	<CALXCt0rJw19OuWKeANe474-rNf_iui6QuTA=RsBePc7f7jvQDQ@mail.gmail.com>
	<1b4e7a32a02c354175edde1e44247d1a@ctidealer.com.br>
	<60dd418aa8ac524b68e17b1cb912fe9a@ctidealer.com.br>
Message-ID: <6b0b8532a2fe27ce8d2466b34b7f06dc@ctidealer.com.br>

 

Nothing changed this way too... 

Em 15.06.2016 15:45,
eduardo at ctidealer.com.br escreveu: 

> This way? 
> 
>
freeswitch at machine> console loglevel 9 
> freeswitch at machine> fsctl
loglevel 9 
> freeswitch at machine> load mod_gsmopen 
> 
> Em 15.06.2016
15:26, eduardo at ctidealer.com.br escreveu: 
> 
>> How can I put in debug
mode? 
>> 
>> Em 15.06.2016 15:09, Giovanni Maruzzelli escreveu: 
>>

>>> put it in debug mode, and check if the modem expose those
events.
>>> 
>>> I would suspect the modem, as is initialized, do
not.
>>> 
>>> So, maybe you may want to find how to initialize that
modem to make it expose those events...
>>> 
>>> It is not a bug
(obviously if events are exposed we use it), it can be a non supported
(eg, non correctly initialized) modem. 
>>> 
>>> On Wed, Jun 15, 2016 at
7:51 PM, edrperes <eduardo at ctidealer.com.br> wrote:
>>> 
>>>> I opened a
JIRA to this question because i think it is as bug
>>>> 
>>>> I tried
many variantions...
>>>> 
>>>> I put a hub with a 3 ampere supply and
nothing changed, for a single modem
>>>> 
>>>> --
>>>> View this message
in context:
http://freeswitch-users.2379917.n2.nabble.com/GsmOpen-call-control-tp7596221p7596226.html
[1]
>>>> 
>>>> Sent from the freeswitch-users mailing list archive at
Nabble.com.
>>>> 
>>>>
_________________________________________________________________________
>>>>
Professional FreeSWITCH Consulting Services:
>>>>
consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
[2]
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
[3]
>>>> http://confluence.freeswitch.org [4]
>>>>
http://www.cluecon.com [5]
>>>> 
>>>> FreeSWITCH-users mailing list
>>>>
FreeSWITCH-users at lists.freeswitch.org
>>>>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [6]
>>>>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[7]
>>>> http://www.freeswitch.org [3]
>>> 
>>> -- 
>>> 
>>>
Sincerely,
>>> 
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618 
>>>

>>>
_________________________________________________________________________
>>>
Professional FreeSWITCH Consulting Services: 
>>>
consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com [2]
>>>

>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org [3]
>>>
http://confluence.freeswitch.org [4]
>>> http://www.cluecon.com [5]
>>>

>>> FreeSWITCH-users mailing list
>>>
FreeSWITCH-users at lists.freeswitch.org
>>>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [6]
>>>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[7]
>>> http://www.freeswitch.org [3]
> 
>
_________________________________________________________________________
>
Professional FreeSWITCH Consulting Services: 
>
consulting at freeswitch.org
> http://www.freeswitchsolutions.com [2]
> 
>
Official FreeSWITCH Sites
> http://www.freeswitch.org [3]
>
http://confluence.freeswitch.org [4]
> http://www.cluecon.com [5]
> 
>
FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [6]
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[7]
> http://www.freeswitch.org [3]

 

Links:
------
[1]
http://freeswitch-users.2379917.n2.nabble.com/GsmOpen-call-control-tp7596221p7596226.html
[2]
http://www.freeswitchsolutions.com
[3] http://www.freeswitch.org
[4]
http://confluence.freeswitch.org
[5] http://www.cluecon.com
[6]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[7]
http://lists.freeswitch.org/mailman/options/freeswitch-users
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From eduardo at ctidealer.com.br  Wed Jun 15 23:31:45 2016
From: eduardo at ctidealer.com.br (edrperes)
Date: Wed, 15 Jun 2016 12:31:45 -0700 (MST)
Subject: [Freeswitch-users] GsmOpen call control
In-Reply-To: <1466013100593-7596226.post@n2.nabble.com>
References: <1465396074152-7596221.post@n2.nabble.com>
	<1465401044177-7596222.post@n2.nabble.com>
	<1466013100593-7596226.post@n2.nabble.com>
Message-ID: <1466019105129-7596227.post@n2.nabble.com>

Could it be a driver problem?

I installed a driver I have found in internet... fro Huawey e1756 modem...

Without it Windows do not recognize the modem nor dc-unlocker recognize the
modem

The driver seems correct...



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/GsmOpen-call-control-tp7596221p7596227.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From eduardo at ctidealer.com.br  Thu Jun 16 19:26:11 2016
From: eduardo at ctidealer.com.br (edrperes)
Date: Thu, 16 Jun 2016 08:26:11 -0700 (MST)
Subject: [Freeswitch-users] GsmOpen call control
In-Reply-To: <1466019105129-7596227.post@n2.nabble.com>
References: <1465396074152-7596221.post@n2.nabble.com>
	<1465401044177-7596222.post@n2.nabble.com>
	<1466013100593-7596226.post@n2.nabble.com>
	<1466019105129-7596227.post@n2.nabble.com>
Message-ID: <1466090771539-7596228.post@n2.nabble.com>

I made a experiment

I access the gsm modem throught hyperterminal to send AT commands directly
to modem

I typed....................at+cpas
received.................+cpas: 0
                                OK

(I call modem's number from another phone)

I typed....................at+cpas
received.................+cpas: 3 (it's ringing, but doesn't show RING)
                               OK


No "RING" message appears on hyperterminal

Could it be related to this problem?



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/GsmOpen-call-control-tp7596221p7596228.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From kathleen at freeswitch.org  Thu Jun 16 19:56:13 2016
From: kathleen at freeswitch.org (Kathleen King)
Date: Thu, 16 Jun 2016 08:56:13 -0700
Subject: [Freeswitch-users] =?utf-8?q?=5BAttend_All_About_the_API_?=
	=?utf-8?q?=E2=80=93_Use_Code_ClueCon_=26_Receive_20=25_off=5D_Let?=
	=?utf-8?q?=E2=80=99s_Talk_API?=
Message-ID: <CAOStUPY3EcQ2XFuvZ34TRW_CAKXwxz8kHWkTtTKQ_2GV5fgRZA@mail.gmail.com>

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From daveh at beachdognet.com  Thu Jun 16 22:13:02 2016
From: daveh at beachdognet.com (Dave Horton)
Date: Thu, 16 Jun 2016 14:13:02 -0400
Subject: [Freeswitch-users] need help using http_cache + s3
Message-ID: <3AE80F4A-C0C2-45D6-8916-E9F31EA40FBA@beachdognet.com>

I?m having trouble using http_cache to download and play a wave file from my amazon S3 bucket.
I?m following the basic config at https://freeswitch.org/confluence/display/FREESWITCH/mod_http_cache but I must be missing something.

My http_cache.conf.xml looks like this:

???????????
<configuration name="http_cache.conf" description="HTTP GET cache">
  <settings>
    <!-- set to true if you want to enable http:// and https:// formats.  Do not use if mod_httapi is also loaded -->
    <param name="enable-file-formats" value="true"/>
    <param name="max-urls" value="10000"/>
    <param name="location" value="$${cache_dir}"/>
    <param name="default-max-age" value="86400"/>
    <param name="prefetch-thread-count" value="8"/>
    <param name="prefetch-queue-size" value="100"/>
    <!-- absolute path to CA bundle file -->
    <param name="ssl-cacert" value="$${certs_dir}/cacert.pem"/>
    <!-- verify certificates -->
    <param name="ssl-verifypeer" value="true"/>
    <!-- verify host name matches certificate -->
    <param name="ssl-verifyhost" value="true"/>
    <!-- default is 300 seconds, override here -->
    <!--param name="connect-timeout" value="300"/-->
    <!-- default is 300 seconds, override here -->
    <!--param name="download-timeout" value="300"/-->
  </settings>

    <profiles>
    <!-- amazon s3 security credentials -->
    <profile name="s3">
      <aws-s3>
        <!-- 20 character key identifier -->
        <access-key-id><![CDATA[my-access-key-here]]></access-key-id>
        <!-- 40 character secret -->
        <secret-access-key><![CDATA[my-secret-access-key-here]]></secret-access-key>
      </aws-s3>
      <!-- optional list of domains that this profile will automatically be applied to -->
      <!-- if you wish to apply the s3 credentials to a domain not listed here, then use
           {profile=s3}http://foo.s3... -->
      <domains>
        <domain name=?my-bucket.s3.amazonaws.com"/>
      </domains>
    </profile>
   </profiles>
</configuration>
?????????????

My dialplan looks like this:

??????????????
   <extension name="announcement">
      <condition field="${sip_req_user}" expression="^annc$"/>
      <condition field="${sip_req_params}" expression="^play=(.*)$">
        <action application="log" data=?file to play: $1"/>
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="playback" data="{profile=s3}http_cache://https://my-bucket.s3.amazonaws.com:/$1"/>
        <action application="hangup"/>
      </condition>
   </extension>
_______________________

The log file says this:

   EXECUTE sofia/drachtio_mrf/5083084809 at 68.64.80.16 playback({profile=s3}http_cache://http://wcs-icr.s3.amazonaws.com:/RBT-5762afa9991351886b295fbc/prompt/1466086249630-call-blocked.wav)
   2016-06-16 14:02:59.173718 [DEBUG] mod_http_cache.c:569 Locked cache
   2016-06-16 14:02:59.173718 [INFO] mod_http_cache.c:661 Cache MISS: size = 0 (0 MB), hit ratio = 0/1
   2016-06-16 14:02:59.173718 [DEBUG] mod_http_cache.c:726 Adding http://wcs-icr.s3.amazonaws.com:/RBT-5762afa9991351886b295fbc/prompt/1466086249630-call-blocked.wav(/usr/local/freeswitch/cache/b6/456219-e1eb-43c2-bbd9-449cfb3af6ac.wav) to cache index 0
   2016-06-16 14:02:59.173718 [DEBUG] mod_http_cache.c:580 Unlocked cache
   2016-06-16 14:02:59.173718 [DEBUG] mod_http_cache.c:1077 opening /usr/local/freeswitch/cache/b6/456219-e1eb-43c2-bbd9-449cfb3af6ac.wav for URL cache
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 HTTP/1.1 403 Forbidden
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 x-amz-request-id: F27F3491740AC96E
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 x-amz-id-2: 1LII609UCqdosrj+3tha2MrWfkoDjIgJ/wK2V47LDkgpn6/i8V4GFRxVzzoXRj9Rzbq6BMP15Po=
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 Content-Type: application/xml
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 Transfer-Encoding: chunked
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 Date: Thu, 16 Jun 2016 18:02:58 GMT
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 Server: AmazonS3
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 
   2016-06-16 14:02:59.233717 [ERR] mod_http_cache.c:1131 Received HTTP error 403 trying to fetch http://wcs-icr.s3.amazonaws.com:/RBT-5762afa9991351886b295fbc/prompt/1466086249630-call-blocked.wav
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:569 Locked cache
   2016-06-16 14:02:59.233717 [INFO] mod_http_cache.c:683 Failed to download URL http://wcs-icr.s3.amazonaws.com:/RBT-5762afa9991351886b295fbc/prompt/1466086249630-call-blocked.wav
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:580 Unlocked cache
   EXECUTE sofia/drachtio_mrf/5083084809 at 68.64.80.16 hangup()


????????

I?ve fiddled with a bunch of different things, but no luck.  I took a tshark of the outgoing http get and I can see it is not putting in the headers to authenticate, so it seems it is not getting into 
the S3 code in http_cache.  The docs do not include a complete S3 working example, unfortunately.

Any help or pointers from those who have gotten this working would be appreciated?

Dave

From ssinyagin at gmail.com  Thu Jun 16 22:13:59 2016
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Thu, 16 Jun 2016 20:13:59 +0200
Subject: [Freeswitch-users] How to playback background call process
In-Reply-To: <00b801d1c715$76ef7420$64ce5c60$@gmail.com>
References: <000701d1c64f$65ff6f10$31fe4d30$@gmail.com>
	<CACJ9VGNY-hU59+S2mFzGeLO6YhHg7c53bjite6JTZXRQe3Bd9Q@mail.gmail.com>
	<00b801d1c715$76ef7420$64ce5c60$@gmail.com>
Message-ID: <CACOYK=jaqMHsijwMjJTpUA-tcnCK+22822DUo5evgco6hZikyw@mail.gmail.com>

You need to reproduce it with latest FreeSWITCH, and file a jira if the
problem persists.
On Jun 15, 2016 16:53, "Tanim" <tanim05 at gmail.com> wrote:

> Thanks for your reply. Its worked partially.
>
> When FS get ringback from far end then its working fine. But if a call
> take more time to get RBT then FS gets crashed. Here is the log:
>
>
>
> [DEBUG] switch_ivr_originate.c:1305 Play Ringback File
> [file_string://digits/youhave.wav!digits/9.wav]
>
> [DEBUG] switch_rtp.c:1846 rtcp_stats_init: audio ssrc[37546101] base_seq[0]
>
> [DEBUG] switch_rtp.c:6707 Correct audio ip/port confirmed.
>
> freeswitch: src/switch_core_file.c:378: switch_core_file_read: Assertion
> `fh->file_interface != ((void *)0)' failed.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *???????
> ???????
> *Sent:* Wednesday, June 15, 2016 6:51 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] How to playback background call process
>
>
>
> Hi
>
> Try to use <action application="set" data="ringback=/IVR/FILE" />
>
>
>
>
>
> 2016-06-14 18:14 GMT+03:00 Tanim <tanim05 at gmail.com>:
>
> Hi,
>
>
>
> I want to playback file background call process. So that in playing IVR
> time, call will process to send to gateways. Currently my dialplan like
> this:
>
> <action application="playback" data="/IVR/FILE" />
>
> <action application="bridge" data="sofia/gateway/gw1/123456" />
>
>
>
> In this way before call processing IVR is playing then start to process
> the call. But I want both in same time.
>
> Can you please help me, how can I do it.
>
>
>
> Thanks in advance.
>
> Tanim
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
>
> WBR
>
> Alex
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From cmrienzo at gmail.com  Thu Jun 16 22:42:39 2016
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Thu, 16 Jun 2016 14:42:39 -0400
Subject: [Freeswitch-users] need help using http_cache + s3
In-Reply-To: <3AE80F4A-C0C2-45D6-8916-E9F31EA40FBA@beachdognet.com>
References: <3AE80F4A-C0C2-45D6-8916-E9F31EA40FBA@beachdognet.com>
Message-ID: <CAD-qRkgew2EdTfNir50s3wyNG5moSz9N6yDhgABe3+xessu+dA@mail.gmail.com>

Given the example configuration, you can remove {profile=s3}http_cache://
from your playback URL.  Also, you can change the config / reload the
module / and test using fs_cli http_get for rapid verification of your
setup.


On Thu, Jun 16, 2016 at 2:13 PM, Dave Horton <daveh at beachdognet.com> wrote:

> I?m having trouble using http_cache to download and play a wave file from
> my amazon S3 bucket.
> I?m following the basic config at
> https://freeswitch.org/confluence/display/FREESWITCH/mod_http_cache but I
> must be missing something.
>
> My http_cache.conf.xml looks like this:
>
> ???????????
> <configuration name="http_cache.conf" description="HTTP GET cache">
>   <settings>
>     <!-- set to true if you want to enable http:// and https:// formats.
> Do not use if mod_httapi is also loaded -->
>     <param name="enable-file-formats" value="true"/>
>     <param name="max-urls" value="10000"/>
>     <param name="location" value="$${cache_dir}"/>
>     <param name="default-max-age" value="86400"/>
>     <param name="prefetch-thread-count" value="8"/>
>     <param name="prefetch-queue-size" value="100"/>
>     <!-- absolute path to CA bundle file -->
>     <param name="ssl-cacert" value="$${certs_dir}/cacert.pem"/>
>     <!-- verify certificates -->
>     <param name="ssl-verifypeer" value="true"/>
>     <!-- verify host name matches certificate -->
>     <param name="ssl-verifyhost" value="true"/>
>     <!-- default is 300 seconds, override here -->
>     <!--param name="connect-timeout" value="300"/-->
>     <!-- default is 300 seconds, override here -->
>     <!--param name="download-timeout" value="300"/-->
>   </settings>
>
>     <profiles>
>     <!-- amazon s3 security credentials -->
>     <profile name="s3">
>       <aws-s3>
>         <!-- 20 character key identifier -->
>         <access-key-id><![CDATA[my-access-key-here]]></access-key-id>
>         <!-- 40 character secret -->
>
> <secret-access-key><![CDATA[my-secret-access-key-here]]></secret-access-key>
>       </aws-s3>
>       <!-- optional list of domains that this profile will automatically
> be applied to -->
>       <!-- if you wish to apply the s3 credentials to a domain not listed
> here, then use
>            {profile=s3}http://foo.s3... -->
>       <domains>
>         <domain name=?my-bucket.s3.amazonaws.com"/>
>       </domains>
>     </profile>
>    </profiles>
> </configuration>
> ?????????????
>
> My dialplan looks like this:
>
> ??????????????
>    <extension name="announcement">
>       <condition field="${sip_req_user}" expression="^annc$"/>
>       <condition field="${sip_req_params}" expression="^play=(.*)$">
>         <action application="log" data=?file to play: $1"/>
>         <action application="answer"/>
>         <action application="sleep" data="1000"/>
>         <action application="playback"
> data="{profile=s3}http_cache://https://my-bucket.s3.amazonaws.com:/$1"/>
>         <action application="hangup"/>
>       </condition>
>    </extension>
> _______________________
>
> The log file says this:
>
>    EXECUTE sofia/drachtio_mrf/5083084809 at 68.64.80.16
> playback({profile=s3}http_cache://http://wcs-icr.s3.amazonaws.com:
> /RBT-5762afa9991351886b295fbc/prompt/1466086249630-call-blocked.wav)
>    2016-06-16 14:02:59.173718 [DEBUG] mod_http_cache.c:569 Locked cache
>    2016-06-16 14:02:59.173718 [INFO] mod_http_cache.c:661 Cache MISS: size
> = 0 (0 MB), hit ratio = 0/1
>    2016-06-16 14:02:59.173718 [DEBUG] mod_http_cache.c:726 Adding
> http://wcs-icr.s3.amazonaws.com:/RBT-5762afa9991351886b295fbc/prompt/1466086249630-call-blocked.wav(/usr/local/freeswitch/cache/b6/456219-e1eb-43c2-bbd9-449cfb3af6ac.wav)
> to cache index 0
>    2016-06-16 14:02:59.173718 [DEBUG] mod_http_cache.c:580 Unlocked cache
>    2016-06-16 14:02:59.173718 [DEBUG] mod_http_cache.c:1077 opening
> /usr/local/freeswitch/cache/b6/456219-e1eb-43c2-bbd9-449cfb3af6ac.wav for
> URL cache
>    2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 HTTP/1.1 403
> Forbidden
>    2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548
> x-amz-request-id: F27F3491740AC96E
>    2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 x-amz-id-2:
> 1LII609UCqdosrj+3tha2MrWfkoDjIgJ/wK2V47LDkgpn6/i8V4GFRxVzzoXRj9Rzbq6BMP15Po=
>    2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 Content-Type:
> application/xml
>    2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548
> Transfer-Encoding: chunked
>    2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 Date: Thu, 16
> Jun 2016 18:02:58 GMT
>    2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 Server: AmazonS3
>    2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548
>    2016-06-16 14:02:59.233717 [ERR] mod_http_cache.c:1131 Received HTTP
> error 403 trying to fetch http://wcs-icr.s3.amazonaws.com:
> /RBT-5762afa9991351886b295fbc/prompt/1466086249630-call-blocked.wav
>    2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:569 Locked cache
>    2016-06-16 14:02:59.233717 [INFO] mod_http_cache.c:683 Failed to
> download URL http://wcs-icr.s3.amazonaws.com:
> /RBT-5762afa9991351886b295fbc/prompt/1466086249630-call-blocked.wav
>    2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:580 Unlocked cache
>    EXECUTE sofia/drachtio_mrf/5083084809 at 68.64.80.16 hangup()
>
>
> ????????
>
> I?ve fiddled with a bunch of different things, but no luck.  I took a
> tshark of the outgoing http get and I can see it is not putting in the
> headers to authenticate, so it seems it is not getting into
> the S3 code in http_cache.  The docs do not include a complete S3 working
> example, unfortunately.
>
> Any help or pointers from those who have gotten this working would be
> appreciated?
>
> Dave
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From daveh at beachdognet.com  Thu Jun 16 23:32:29 2016
From: daveh at beachdognet.com (Dave Horton)
Date: Thu, 16 Jun 2016 15:32:29 -0400
Subject: [Freeswitch-users] need help using http_cache + s3
In-Reply-To: <CAD-qRkgew2EdTfNir50s3wyNG5moSz9N6yDhgABe3+xessu+dA@mail.gmail.com>
References: <3AE80F4A-C0C2-45D6-8916-E9F31EA40FBA@beachdognet.com>
	<CAD-qRkgew2EdTfNir50s3wyNG5moSz9N6yDhgABe3+xessu+dA@mail.gmail.com>
Message-ID: <2ED3C2A5-9CB0-4E98-8260-973819E3801E@beachdognet.com>

Well, from looking at the code, part of the problem is that there is an additional ? undocumented ? configuration parameter called ?base-domain? which is required for the AWS stuff to kick in?.still trying to figure out how this works, by reading the code??anyone who has knowledge please chime in


On Jun 16, 2016, at 2:42 PM, Christopher Rienzo <cmrienzo at gmail.com> wrote:

Given the example configuration, you can remove {profile=s3}http_cache:// from your playback URL.  Also, you can change the config / reload the module / and test using fs_cli http_get for rapid verification of your setup.


On Thu, Jun 16, 2016 at 2:13 PM, Dave Horton <daveh at beachdognet.com <mailto:daveh at beachdognet.com>> wrote:
I?m having trouble using http_cache to download and play a wave file from my amazon S3 bucket.
I?m following the basic config at https://freeswitch.org/confluence/display/FREESWITCH/mod_http_cache <https://freeswitch.org/confluence/display/FREESWITCH/mod_http_cache> but I must be missing something.

My http_cache.conf.xml looks like this:

???????????
<configuration name="http_cache.conf" description="HTTP GET cache">
  <settings>
    <!-- set to true if you want to enable http:// and https:// formats.  Do not use if mod_httapi is also loaded -->
    <param name="enable-file-formats" value="true"/>
    <param name="max-urls" value="10000"/>
    <param name="location" value="$${cache_dir}"/>
    <param name="default-max-age" value="86400"/>
    <param name="prefetch-thread-count" value="8"/>
    <param name="prefetch-queue-size" value="100"/>
    <!-- absolute path to CA bundle file -->
    <param name="ssl-cacert" value="$${certs_dir}/cacert.pem"/>
    <!-- verify certificates -->
    <param name="ssl-verifypeer" value="true"/>
    <!-- verify host name matches certificate -->
    <param name="ssl-verifyhost" value="true"/>
    <!-- default is 300 seconds, override here -->
    <!--param name="connect-timeout" value="300"/-->
    <!-- default is 300 seconds, override here -->
    <!--param name="download-timeout" value="300"/-->
  </settings>

    <profiles>
    <!-- amazon s3 security credentials -->
    <profile name="s3">
      <aws-s3>
        <!-- 20 character key identifier -->
        <access-key-id><![CDATA[my-access-key-here]]></access-key-id>
        <!-- 40 character secret -->
        <secret-access-key><![CDATA[my-secret-access-key-here]]></secret-access-key>
      </aws-s3>
      <!-- optional list of domains that this profile will automatically be applied to -->
      <!-- if you wish to apply the s3 credentials to a domain not listed here, then use
           {profile=s3}http://foo.s3. <http://foo.s3./>.. -->
      <domains>
        <domain name=?my-bucket.s3.amazonaws.com <http://my-bucket.s3.amazonaws.com/>"/>
      </domains>
    </profile>
   </profiles>
</configuration>
?????????????

My dialplan looks like this:

??????????????
   <extension name="announcement">
      <condition field="${sip_req_user}" expression="^annc$"/>
      <condition field="${sip_req_params}" expression="^play=(.*)$">
        <action application="log" data=?file to play: $1"/>
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="playback" data="{profile=s3}http_cache://https://my-bucket.s3.amazonaws.com:/$1"/>
        <action application="hangup"/>
      </condition>
   </extension>
_______________________

The log file says this:

   EXECUTE sofia/drachtio_mrf/5083084809 at 68.64.80.16 <mailto:5083084809 at 68.64.80.16> playback({profile=s3}http_cache://http://wcs-icr.s3.amazonaws.com:/RBT-5762afa9991351886b295fbc/prompt/1466086249630-call-blocked.wav)
   2016-06-16 14:02:59.173718 [DEBUG] mod_http_cache.c:569 Locked cache
   2016-06-16 14:02:59.173718 [INFO] mod_http_cache.c:661 Cache MISS: size = 0 (0 MB), hit ratio = 0/1
   2016-06-16 14:02:59.173718 [DEBUG] mod_http_cache.c:726 Adding http://wcs-icr.s3.amazonaws.com:/RBT-5762afa9991351886b295fbc/prompt/1466086249630-call-blocked.wav(/usr/local/freeswitch/cache/b6/456219-e1eb-43c2-bbd9-449cfb3af6ac.wav) to cache index 0
   2016-06-16 14:02:59.173718 [DEBUG] mod_http_cache.c:580 Unlocked cache
   2016-06-16 14:02:59.173718 [DEBUG] mod_http_cache.c:1077 opening /usr/local/freeswitch/cache/b6/456219-e1eb-43c2-bbd9-449cfb3af6ac.wav for URL cache
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 HTTP/1.1 403 Forbidden
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 x-amz-request-id: F27F3491740AC96E
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 x-amz-id-2: 1LII609UCqdosrj+3tha2MrWfkoDjIgJ/wK2V47LDkgpn6/i8V4GFRxVzzoXRj9Rzbq6BMP15Po=
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 Content-Type: application/xml
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 Transfer-Encoding: chunked
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 Date: Thu, 16 Jun 2016 18:02:58 GMT
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 Server: AmazonS3
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548
   2016-06-16 14:02:59.233717 [ERR] mod_http_cache.c:1131 Received HTTP error 403 trying to fetch http://wcs-icr.s3.amazonaws.com:/RBT-5762afa9991351886b295fbc/prompt/1466086249630-call-blocked.wav
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:569 Locked cache
   2016-06-16 14:02:59.233717 [INFO] mod_http_cache.c:683 Failed to download URL http://wcs-icr.s3.amazonaws.com:/RBT-5762afa9991351886b295fbc/prompt/1466086249630-call-blocked.wav
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:580 Unlocked cache
   EXECUTE sofia/drachtio_mrf/5083084809 at 68.64.80.16 <mailto:5083084809 at 68.64.80.16> hangup()


????????

I?ve fiddled with a bunch of different things, but no luck.  I took a tshark of the outgoing http get and I can see it is not putting in the headers to authenticate, so it seems it is not getting into
the S3 code in http_cache.  The docs do not include a complete S3 working example, unfortunately.

Any help or pointers from those who have gotten this working would be appreciated?

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

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

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_________________________________________________________________________
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consulting at freeswitch.org
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Official FreeSWITCH Sites
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From cmrienzo at gmail.com  Thu Jun 16 23:42:00 2016
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Thu, 16 Jun 2016 15:42:00 -0400
Subject: [Freeswitch-users] need help using http_cache + s3
In-Reply-To: <2ED3C2A5-9CB0-4E98-8260-973819E3801E@beachdognet.com>
References: <3AE80F4A-C0C2-45D6-8916-E9F31EA40FBA@beachdognet.com>
	<CAD-qRkgew2EdTfNir50s3wyNG5moSz9N6yDhgABe3+xessu+dA@mail.gmail.com>
	<2ED3C2A5-9CB0-4E98-8260-973819E3801E@beachdognet.com>
Message-ID: <CAD-qRkjsj_Nx-9k7ppJ3R-TVBt0UpjAOnYfMtzRJgn7KSJY96w@mail.gmail.com>

Base-domain stuff is only if you host your own S3-compatible service.  It
will default to amazonaws.com if not set.  If setting profile=s3 doesn't
work, then something is really broken in the config.

On Thu, Jun 16, 2016 at 3:32 PM, Dave Horton <daveh at beachdognet.com> wrote:

> Well, from looking at the code, part of the problem is that there is an
> additional ? undocumented ? configuration parameter called ?base-domain?
> which is required for the AWS stuff to kick in?.still trying to figure out
> how this works, by reading the code??anyone who has knowledge please chime
> in
>
>
> On Jun 16, 2016, at 2:42 PM, Christopher Rienzo <cmrienzo at gmail.com>
> wrote:
>
> Given the example configuration, you can remove {profile=s3}http_cache://
> from your playback URL.  Also, you can change the config / reload the
> module / and test using fs_cli http_get for rapid verification of your
> setup.
>
>
> On Thu, Jun 16, 2016 at 2:13 PM, Dave Horton <daveh at beachdognet.com>
> wrote:
>
>> I?m having trouble using http_cache to download and play a wave file from
>> my amazon S3 bucket.
>> I?m following the basic config at
>> https://freeswitch.org/confluence/display/FREESWITCH/mod_http_cache but
>> I must be missing something.
>>
>> My http_cache.conf.xml looks like this:
>>
>> ???????????
>> <configuration name="http_cache.conf" description="HTTP GET cache">
>>   <settings>
>>     <!-- set to true if you want to enable http:// and https://
>> formats.  Do not use if mod_httapi is also loaded -->
>>     <param name="enable-file-formats" value="true"/>
>>     <param name="max-urls" value="10000"/>
>>     <param name="location" value="$${cache_dir}"/>
>>     <param name="default-max-age" value="86400"/>
>>     <param name="prefetch-thread-count" value="8"/>
>>     <param name="prefetch-queue-size" value="100"/>
>>     <!-- absolute path to CA bundle file -->
>>     <param name="ssl-cacert" value="$${certs_dir}/cacert.pem"/>
>>     <!-- verify certificates -->
>>     <param name="ssl-verifypeer" value="true"/>
>>     <!-- verify host name matches certificate -->
>>     <param name="ssl-verifyhost" value="true"/>
>>     <!-- default is 300 seconds, override here -->
>>     <!--param name="connect-timeout" value="300"/-->
>>     <!-- default is 300 seconds, override here -->
>>     <!--param name="download-timeout" value="300"/-->
>>   </settings>
>>
>>     <profiles>
>>     <!-- amazon s3 security credentials -->
>>     <profile name="s3">
>>       <aws-s3>
>>         <!-- 20 character key identifier -->
>>         <access-key-id><![CDATA[my-access-key-here]]></access-key-id>
>>         <!-- 40 character secret -->
>>
>> <secret-access-key><![CDATA[my-secret-access-key-here]]></secret-access-key>
>>       </aws-s3>
>>       <!-- optional list of domains that this profile will automatically
>> be applied to -->
>>       <!-- if you wish to apply the s3 credentials to a domain not listed
>> here, then use
>>            {profile=s3}http://foo.s3... -->
>>       <domains>
>>         <domain name=?my-bucket.s3.amazonaws.com"/>
>>       </domains>
>>     </profile>
>>    </profiles>
>> </configuration>
>> ?????????????
>>
>> My dialplan looks like this:
>>
>> ??????????????
>>    <extension name="announcement">
>>       <condition field="${sip_req_user}" expression="^annc$"/>
>>       <condition field="${sip_req_params}" expression="^play=(.*)$">
>>         <action application="log" data=?file to play: $1"/>
>>         <action application="answer"/>
>>         <action application="sleep" data="1000"/>
>>         <action application="playback" data="{profile=s3}http_cache://
>> https://my-bucket.s3.amazonaws.com:/$1"/>
>>         <action application="hangup"/>
>>       </condition>
>>    </extension>
>> _______________________
>>
>> The log file says this:
>>
>>    EXECUTE sofia/drachtio_mrf/5083084809 at 68.64.80.16
>> playback({profile=s3}http_cache://
>> http://wcs-icr.s3.amazonaws.com:/RBT-5762afa9991351886b295fbc/prompt/1466086249630-call-blocked.wav
>> )
>>    2016-06-16 14:02:59.173718 [DEBUG] mod_http_cache.c:569 Locked cache
>>    2016-06-16 14:02:59.173718 [INFO] mod_http_cache.c:661 Cache MISS:
>> size = 0 (0 MB), hit ratio = 0/1
>>    2016-06-16 14:02:59.173718 [DEBUG] mod_http_cache.c:726 Adding
>> http://wcs-icr.s3.amazonaws.com:/RBT-5762afa9991351886b295fbc/prompt/1466086249630-call-blocked.wav(/usr/local/freeswitch/cache/b6/456219-e1eb-43c2-bbd9-449cfb3af6ac.wav)
>> to cache index 0
>>    2016-06-16 14:02:59.173718 [DEBUG] mod_http_cache.c:580 Unlocked cache
>>    2016-06-16 14:02:59.173718 [DEBUG] mod_http_cache.c:1077 opening
>> /usr/local/freeswitch/cache/b6/456219-e1eb-43c2-bbd9-449cfb3af6ac.wav for
>> URL cache
>>    2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 HTTP/1.1 403
>> Forbidden
>>    2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548
>> x-amz-request-id: F27F3491740AC96E
>>    2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 x-amz-id-2:
>> 1LII609UCqdosrj+3tha2MrWfkoDjIgJ/wK2V47LDkgpn6/i8V4GFRxVzzoXRj9Rzbq6BMP15Po=
>>    2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 Content-Type:
>> application/xml
>>    2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548
>> Transfer-Encoding: chunked
>>    2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 Date: Thu, 16
>> Jun 2016 18:02:58 GMT
>>    2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 Server:
>> AmazonS3
>>    2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548
>>    2016-06-16 14:02:59.233717 [ERR] mod_http_cache.c:1131 Received HTTP
>> error 403 trying to fetch
>> http://wcs-icr.s3.amazonaws.com:/RBT-5762afa9991351886b295fbc/prompt/1466086249630-call-blocked.wav
>>    2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:569 Locked cache
>>    2016-06-16 14:02:59.233717 [INFO] mod_http_cache.c:683 Failed to
>> download URL
>> http://wcs-icr.s3.amazonaws.com:/RBT-5762afa9991351886b295fbc/prompt/1466086249630-call-blocked.wav
>>    2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:580 Unlocked cache
>>    EXECUTE sofia/drachtio_mrf/5083084809 at 68.64.80.16 hangup()
>>
>>
>> ????????
>>
>> I?ve fiddled with a bunch of different things, but no luck.  I took a
>> tshark of the outgoing http get and I can see it is not putting in the
>> headers to authenticate, so it seems it is not getting into
>> the S3 code in http_cache.  The docs do not include a complete S3 working
>> example, unfortunately.
>>
>> Any help or pointers from those who have gotten this working would be
>> appreciated?
>>
>> Dave
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From schoch+freeswitch.org at xwin32.com  Thu Jun 16 23:43:06 2016
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Thu, 16 Jun 2016 12:43:06 -0700
Subject: [Freeswitch-users] Run a LUA script when voicemail received
In-Reply-To: <CAOkuKdf_u0h8pOQSJBBntFNaKr0OeFseyEmTOuDiBrJZpimt0A@mail.gmail.com>
References: <CAOkuKdfnwFsKN4H7ZxB+dvfGSGnAby59AM+kO4MCrLuJWa=Qag@mail.gmail.com>
	<CAEwTsAg0rPXMtvJtGns1kwcNWRT2rHJSnn+psXfnF9_BMdV3Ww@mail.gmail.com>
	<CAFmZFiNHye2Tujfwsw8mh9tc+bnZNxSbxTt670W0L3UtCKszGg@mail.gmail.com>
	<CAOkuKdczNFzL76oRF1f2SARJ7700owGmMnJOrtTZA6x=5AsLJQ@mail.gmail.com>
	<CAOkuKde_1pK6TzA4-9qKSVuh8A0s7i1qs2n4uUt-5-b8=HSgKQ@mail.gmail.com>
	<CAOkuKdcvRDVkmOhdMQ+7VB8edhdbzMX+25-wRAxgPSn-waX4iw@mail.gmail.com>
	<CAOkuKddVsmLA-+YjQo7myFF7HXta6K468t+7WooETYU6=_vJkw@mail.gmail.com>
	<CAFmZFiN+q-MgFr6PfZEnDVe9hvqikASGwvCuS4A4orvX57RS7Q@mail.gmail.com>
	<CAOkuKdf_u0h8pOQSJBBntFNaKr0OeFseyEmTOuDiBrJZpimt0A@mail.gmail.com>
Message-ID: <CAOkuKdenNcaV4WhHB_6ha_KkwcPVSEd=0fWCUzpZB7b9_L9Pdw@mail.gmail.com>

In browsing through the source to mod_curl, I found an option that allows
me to use curl without invoking a shell. I made some minor changes to the
script, changing the url to this:

url = "https://user:password at api.flowroute.com/v2/messages"

... adding some escaping after the message is composed...

message = message .. "."
-- Add JSON string escapes to the message
message = string.gsub(message, "([\\\"'])", "\\%1")

...and using the mod_curl API call to send the text to Flowroute:

api:execute("curl", url .. " content-type application/json post '" .. data
.. "'")

Works, easy, efficient.

-- 
Steve


On Thu, Jun 2, 2016 at 3:26 PM, Steven Schoch <
schoch+freeswitch.org at xwin32.com> wrote:

> I got it working! The goal was to run a script to inform users via a text
> message when they get a voicemail.
> This script looks for a "vm-notify-sms" param in the user's directory
> information. It sends the text from user's outbound caller-ID number, which
> is set in the standard FS configuration. I'm using the SMS API from
> Flowroute.com.
>
> --[[ This script is called by configuration in
> autoload_configs/lua.conf.xml.
>      It is called on a "vm:maintenance" event.
> ]]
>
> -- The following is specifc to the Flowroute SMS API,
> -- documented at https://developer.flowroute.com/docs/messaging
> url = "https://api.flowroute.com/v2/messages"
> authentication = "user:password"
>
> action = event:getHeader("VM-Action")
> if action ~= "leave-message" then return end
> user = event:getHeader("VM-User")
> domain = event:getHeader("VM-Domain")
> caller_id_name = event:getHeader("VM-Caller-ID-Name")
> caller_id_number = event:getHeader("VM-Caller-ID-Number")
> --[[ We don't care about these:
> file_path = event:getHeader("VM-File-Path")
> flags = event:getHeader("VM-Flags")
> folder = event:getHeader("VM-Folder")
> uuid = event:getHeader("VM-UUID")
> --]]
> message_len = event:getHeader("VM-Message-Len")
> timestamp = event:getHeader("VM-Timestamp")
> api = freeswitch.API();
> from = api:execute("user_data", user .. "@" .. domain ..  " var
> outbound_caller_id_number")
> to = api:execute("user_data", user .. "@" .. domain .. " param
> vm-notify-sms")
> if to == "" then return end
>
> if caller_id_name == ''
>         or caller_id_name == 'UNKNOWN'
>         or caller_id_name == 'UNASSIGNED'
>         or caller_id_name == 'WIRELESS CALLER'
>         or caller_id_name == 'TOLL FREE CALL'
>         or caller_id_name == 'Anonymous'
>         or caller_id_name == 'Unavailable'
>         then caller_id_name = nil end
> if caller_id_number == ''
>         then caller_id_number = nil end
> message = "Voicemail "
> if caller_id_name
>         then message = message .. "from " .. caller_id_name .. " (" ..
>  caller_id_number .. ")"
>         elseif caller_id_number
>         then message = message .. "from " .. caller_id_number end
> message = message .. " at " .. os.date("%I:%M %p", timestamp)
> message = message .. " length " .. message_len .. " seconds"
> message = message .. " to box " .. user
> message = message .. "."
> -- Blindly send the text
> data = '{ "to": "' .. to .. '", "from": "' .. from .. '", "body": "' ..
> message .. '"}'
> --[[
>      Until either Freeswitch has the ability to pass authentication data
>      to the mod_curl API (FS-9223), or the Flowroute API has the ability
>      to take authentication data in the POST body (they are working on it),
>      we are forced to spawn a shell to use the curl from our OS.
> ]]
> -- api.execute("curl", url .. " auth " .. authentication .. " post " ..
> data)
> os.execute("curl -u " .. authentication .. " -H 'Content-Type:
> application/json' -X POST -d '" .. data .. "' "  .. url)
>
>
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From danny.gershman at gmail.com  Fri Jun 17 02:54:52 2016
From: danny.gershman at gmail.com (Danny Gershman)
Date: Thu, 16 Jun 2016 22:54:52 +0000
Subject: [Freeswitch-users] conference play video
Message-ID: <CAEorZymOmq5b88qBgvCEO5-eoucLx99Si-cYu4HNKi4-Su16fg@mail.gmail.com>

I'm having an issue playing back video in a conference, running (1.6.8 tag
and loaded up mod_av), which it only plays audio back (no visual).  I'm
dialed in using a softphone with VP8 codec.

I run command:

conference <conf> play http://example.org/play.mp4

Any pointers.

Thanks,
Danny Gershman
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From mike at jerris.com  Fri Jun 17 04:07:10 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 16 Jun 2016 20:07:10 -0400
Subject: [Freeswitch-users] conference play video
In-Reply-To: <CAEorZymOmq5b88qBgvCEO5-eoucLx99Si-cYu4HNKi4-Su16fg@mail.gmail.com>
References: <CAEorZymOmq5b88qBgvCEO5-eoucLx99Si-cYu4HNKi4-Su16fg@mail.gmail.com>
Message-ID: <618BE480-F3CD-4903-85A5-923FD12C2941@jerris.com>

Try 1.6.9 as thats now released, if thats not working, take a look at the debug logs and see what it says.  One common issue is playing mp4 files requires the libav-extras package on debian.


> On Jun 16, 2016, at 6:54 PM, Danny Gershman <danny.gershman at gmail.com> wrote:
> 
> I'm having an issue playing back video in a conference, running (1.6.8 tag and loaded up mod_av), which it only plays audio back (no visual).  I'm dialed in using a softphone with VP8 codec.
> 
> I run command:
> 
> conference <conf> play http://example.org/play.mp4 <http://example.org/play.mp4>
> 
> Any pointers.
> 
> Thanks,
> Danny Gershman
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From daveh at beachdognet.com  Fri Jun 17 05:29:15 2016
From: daveh at beachdognet.com (Dave Horton)
Date: Thu, 16 Jun 2016 21:29:15 -0400
Subject: [Freeswitch-users] need help using http_cache + s3
In-Reply-To: <CAD-qRkjsj_Nx-9k7ppJ3R-TVBt0UpjAOnYfMtzRJgn7KSJY96w@mail.gmail.com>
References: <3AE80F4A-C0C2-45D6-8916-E9F31EA40FBA@beachdognet.com>
	<CAD-qRkgew2EdTfNir50s3wyNG5moSz9N6yDhgABe3+xessu+dA@mail.gmail.com>
	<2ED3C2A5-9CB0-4E98-8260-973819E3801E@beachdognet.com>
	<CAD-qRkjsj_Nx-9k7ppJ3R-TVBt0UpjAOnYfMtzRJgn7KSJY96w@mail.gmail.com>
Message-ID: <6BD536A7-10AF-4313-9F30-15E0B67F9DE8@beachdognet.com>

OK, I got it working - tweaked a bunch of things so not sure what was the original problem, but for posterity here is the working config:

??http_cache.conf.xml?

    <profiles>
    <!-- amazon s3 security credentials -->
    <profile name="s3">
      <aws-s3>
        <!-- 20 character key identifier -->
        <access-key-id><![CDATA[my-access-key-id]]></access-key-id>
        <!-- 40 character secret -->
        <secret-access-key><![my-secret-access-key]]></secret-access-key>
      </aws-s3>

      <!-- optional list of domains that this profile will automatically be applied to -->
      <!-- if you wish to apply the s3 credentials to a domain not listed here, then use
           {profile=s3}http://foo.s3... -->
      <domains>
        <domain name=?my-bucket.s3.amazonaws.com"/>
      </domains>
?????????

???dialplan???

   <extension name="announcement">
      <condition field="${sip_req_user}" expression="^annc$"/>
      <condition field="${sip_req_params}" expression="^play=(.*)$">
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="playback" data="http_cache://http://my-bucket.amazonaws.com/$1"/>
        <action application="hangup"/>
      </condition>
   </extension>

------------------------
On Jun 16, 2016, at 3:42 PM, Christopher Rienzo <cmrienzo at gmail.com> wrote:

Base-domain stuff is only if you host your own S3-compatible service.  It will default to amazonaws.com <http://amazonaws.com/> if not set.  If setting profile=s3 doesn't work, then something is really broken in the config.

On Thu, Jun 16, 2016 at 3:32 PM, Dave Horton <daveh at beachdognet.com <mailto:daveh at beachdognet.com>> wrote:
Well, from looking at the code, part of the problem is that there is an additional ? undocumented ? configuration parameter called ?base-domain? which is required for the AWS stuff to kick in?.still trying to figure out how this works, by reading the code??anyone who has knowledge please chime in


On Jun 16, 2016, at 2:42 PM, Christopher Rienzo <cmrienzo at gmail.com <mailto:cmrienzo at gmail.com>> wrote:

Given the example configuration, you can remove {profile=s3}http_cache:// from your playback URL.  Also, you can change the config / reload the module / and test using fs_cli http_get for rapid verification of your setup.


On Thu, Jun 16, 2016 at 2:13 PM, Dave Horton <daveh at beachdognet.com <mailto:daveh at beachdognet.com>> wrote:
I?m having trouble using http_cache to download and play a wave file from my amazon S3 bucket.
I?m following the basic config at https://freeswitch.org/confluence/display/FREESWITCH/mod_http_cache <https://freeswitch.org/confluence/display/FREESWITCH/mod_http_cache> but I must be missing something.

My http_cache.conf.xml looks like this:

???????????
<configuration name="http_cache.conf" description="HTTP GET cache">
  <settings>
    <!-- set to true if you want to enable http:// and https:// formats.  Do not use if mod_httapi is also loaded -->
    <param name="enable-file-formats" value="true"/>
    <param name="max-urls" value="10000"/>
    <param name="location" value="$${cache_dir}"/>
    <param name="default-max-age" value="86400"/>
    <param name="prefetch-thread-count" value="8"/>
    <param name="prefetch-queue-size" value="100"/>
    <!-- absolute path to CA bundle file -->
    <param name="ssl-cacert" value="$${certs_dir}/cacert.pem"/>
    <!-- verify certificates -->
    <param name="ssl-verifypeer" value="true"/>
    <!-- verify host name matches certificate -->
    <param name="ssl-verifyhost" value="true"/>
    <!-- default is 300 seconds, override here -->
    <!--param name="connect-timeout" value="300"/-->
    <!-- default is 300 seconds, override here -->
    <!--param name="download-timeout" value="300"/-->
  </settings>

    <profiles>
    <!-- amazon s3 security credentials -->
    <profile name="s3">
      <aws-s3>
        <!-- 20 character key identifier -->
        <access-key-id><![CDATA[my-access-key-here]]></access-key-id>
        <!-- 40 character secret -->
        <secret-access-key><![CDATA[my-secret-access-key-here]]></secret-access-key>
      </aws-s3>
      <!-- optional list of domains that this profile will automatically be applied to -->
      <!-- if you wish to apply the s3 credentials to a domain not listed here, then use
           {profile=s3}http://foo.s3. <http://foo.s3./>.. -->
      <domains>
        <domain name=?my-bucket.s3.amazonaws.com <http://my-bucket.s3.amazonaws.com/>"/>
      </domains>
    </profile>
   </profiles>
</configuration>
?????????????

My dialplan looks like this:

??????????????
   <extension name="announcement">
      <condition field="${sip_req_user}" expression="^annc$"/>
      <condition field="${sip_req_params}" expression="^play=(.*)$">
        <action application="log" data=?file to play: $1"/>
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="playback" data="{profile=s3}http_cache://https://my-bucket.s3.amazonaws.com:/$1 <https://my-bucket.s3.amazonaws.com/$1>"/>
        <action application="hangup"/>
      </condition>
   </extension>
_______________________

The log file says this:

   EXECUTE sofia/drachtio_mrf/5083084809 at 68.64.80.16 <mailto:5083084809 at 68.64.80.16> playback({profile=s3}http_cache://http://wcs-icr.s3.amazonaws.com:/RBT-5762afa9991351886b295fbc/prompt/1466086249630-call-blocked.wav <http://wcs-icr.s3.amazonaws.com/RBT-5762afa9991351886b295fbc/prompt/1466086249630-call-blocked.wav>)
   2016-06-16 14:02:59.173718 [DEBUG] mod_http_cache.c:569 Locked cache
   2016-06-16 14:02:59.173718 [INFO] mod_http_cache.c:661 Cache MISS: size = 0 (0 MB), hit ratio = 0/1
   2016-06-16 14:02:59.173718 [DEBUG] mod_http_cache.c:726 Adding http://wcs-icr.s3.amazonaws.com:/RBT-5762afa9991351886b295fbc/prompt/1466086249630-call-blocked.wav(/usr/local/freeswitch/cache/b6/456219-e1eb-43c2-bbd9-449cfb3af6ac.wav) <http://wcs-icr.s3.amazonaws.com/RBT-5762afa9991351886b295fbc/prompt/1466086249630-call-blocked.wav(/usr/local/freeswitch/cache/b6/456219-e1eb-43c2-bbd9-449cfb3af6ac.wav)> to cache index 0
   2016-06-16 14:02:59.173718 [DEBUG] mod_http_cache.c:580 Unlocked cache
   2016-06-16 14:02:59.173718 [DEBUG] mod_http_cache.c:1077 opening /usr/local/freeswitch/cache/b6/456219-e1eb-43c2-bbd9-449cfb3af6ac.wav for URL cache
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 HTTP/1.1 403 Forbidden
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 x-amz-request-id: F27F3491740AC96E
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 x-amz-id-2: 1LII609UCqdosrj+3tha2MrWfkoDjIgJ/wK2V47LDkgpn6/i8V4GFRxVzzoXRj9Rzbq6BMP15Po=
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 Content-Type: application/xml
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 Transfer-Encoding: chunked
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 Date: Thu, 16 Jun 2016 18:02:58 GMT
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548 Server: AmazonS3
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:548
   2016-06-16 14:02:59.233717 [ERR] mod_http_cache.c:1131 Received HTTP error 403 trying to fetch http://wcs-icr.s3.amazonaws.com:/RBT-5762afa9991351886b295fbc/prompt/1466086249630-call-blocked.wav <http://wcs-icr.s3.amazonaws.com/RBT-5762afa9991351886b295fbc/prompt/1466086249630-call-blocked.wav>
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:569 Locked cache
   2016-06-16 14:02:59.233717 [INFO] mod_http_cache.c:683 Failed to download URL http://wcs-icr.s3.amazonaws.com:/RBT-5762afa9991351886b295fbc/prompt/1466086249630-call-blocked.wav <http://wcs-icr.s3.amazonaws.com/RBT-5762afa9991351886b295fbc/prompt/1466086249630-call-blocked.wav>
   2016-06-16 14:02:59.233717 [DEBUG] mod_http_cache.c:580 Unlocked cache
   EXECUTE sofia/drachtio_mrf/5083084809 at 68.64.80.16 <mailto:5083084809 at 68.64.80.16> hangup()


????????

I?ve fiddled with a bunch of different things, but no luck.  I took a tshark of the outgoing http get and I can see it is not putting in the headers to authenticate, so it seems it is not getting into
the S3 code in http_cache.  The docs do not include a complete S3 working example, unfortunately.

Any help or pointers from those who have gotten this working would be appreciated?

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

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

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

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

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

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

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

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

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

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

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

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From anthony.minessale at gmail.com  Fri Jun 17 05:32:45 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 16 Jun 2016 20:32:45 -0500
Subject: [Freeswitch-users] If you are doing anything cool with FreeSWITCH
 or want to see what others have done!
Message-ID: <CAKbxfG8wT13DSt2XZiWAcVma3TH5vc7U3ZLZm2n62xSZfVtCoA@mail.gmail.com>

Come see us at ClueCon
https://cluecon.com

Its one of the easiest ways to show support for the project that works so
hard all year and you get a great experience in return!

Call us at 877 7 4A CLUE (877-742-2583) if you want to find out about
promotions or group discounts.

We also may have some free tickets available to those willing to help out.





-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From anthony.minessale at gmail.com  Fri Jun 17 06:04:11 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 16 Jun 2016 21:04:11 -0500
Subject: [Freeswitch-users] 12th Annual Communication Conference Features
 Telephony, IoT, Making, and WebRTC
Message-ID: <CAKbxfG9RCoSOPO-Ktiev6zZ7qHUv+TYMoDAbWTTxwHERAocFjg@mail.gmail.com>

August 8-11th 877-742-2583 cluecon.com
12th Annual Telephony / WebRTC / IOT Conference!

ClueCon was founded in 2005 by the FreeSWITCH developers and continues to
be the best place to learn about Open-Source telecom and it gets bigger
every year.
*Great Talks and Panels!*

As you know, ClueCon features three full days of presentations and panels.
We cover FreeSWITCH, Kamailio, OpenSIPS, Asterisk and so much more. Head
over to Cluecon.com to see the list of speakers we already have lined up
this year!

*Register now using ccsave25 to get 25% off registration!*
*ClueCon 2016 Grand prize!*

*Register before June 24th to get 15 raffle tickets per day and 3 drink
coupons for the Gigabit reception! *The coveted prize at ClueCon is an
engraved MacBook made especially for the conference. This Macbook is given
away during a raffle on the last day of the conference. Raffle tickets are
allotted based on registration dates and registering earlier increases your
chances to win! If you wait, but register before July 8th you will get 5
raffle tickets per day and 1 drink coupon for the Gigabit reception!
*ClueCon Coder Games!*

We added the ClueCon Coder Games last year and its back again for 2016 with
more options including 3D printing and IOT elements! The annual hack-a-thon
gets things started right with a whole day devoted to coding, 3D printing,
and building tech! Be sure to sign-up for free spots in the API and the
Maker challenges! These competitions will pit you against other attendees
for a chance to win some awesome prizes and show off what you can do! You
don't want to miss it!
*Annual Pizza Party!*

The Monday night meet-and-greet pizza party is a ClueCon staple and it?s
back again this year with a twist. We have a new restaurant providing the
pizza this year, Giodano's. Be sure to register early and reserve your
place to eat Chicago style pizza with the authors of all your favorite
Open-Source Telephony projects!

*After hours social events!*

We took your feedback from the last ClueCon and put some real thought in to
our social events this year, including the Gigabit reception and the
ClueCon After Party hosted by Telnyx! Our Gigabit reception and after hours
karaoke last year were a great success with many attendees listing them as
the highlight of the conference! And, this year we have added another night
of fun to the schedule sponsored by the Chicago natives, Telnyx! Register
now to R.S.V.P. for what are sure to be a magnificent evening of games,
drinks, and developers!
*FreeSWITCH and OpenSIPS Trainings!*

There are options available to stay an extra day and get FreeSWITCH
training with the official curriculum drafted by the FreeSWITCH docs team
and approved by the developers. And, this year we have also arranged to
have OpenSIPS training available as well!
Register Today! <https://cluecon.com?FSUL>
<http://www.twitter.com/cluecon>
<http://www.facebook.com/cluecon/>
<https://cluecon.com>
*Copyright ? 2016 FreeSWITCH Solutions, All rights reserved.*


*Our mailing address is:*
FreeSWITCH Solutions 17345 Civic Drive #2531 Brookfield, WI 53045
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From john.nash778 at gmail.com  Fri Jun 17 08:45:48 2016
From: john.nash778 at gmail.com (John Nash)
Date: Fri, 17 Jun 2016 10:15:48 +0530
Subject: [Freeswitch-users] Freeswitch changes payload type of codec
In-Reply-To: <CAA4rGPy1Kp+9G7X=em6arHOCZ=M8i9JZW0c=tJhWySG-5dv=Ow@mail.gmail.com>
References: <CAA4rGPy7nBRKux0ZkrBTiXz8A72cg0U1zJ+Zct+ppjXaMSpDAw@mail.gmail.com>
	<14ed01d1c312$63834530$2a89cf90$@freeswitch.org>
	<CAA4rGPy1Kp+9G7X=em6arHOCZ=M8i9JZW0c=tJhWySG-5dv=Ow@mail.gmail.com>
Message-ID: <CAA4rGPzD-iiMQURYcnk3N37CnNe2ZeRvVGobdbqp8_YisCBN0w@mail.gmail.com>

Looks like it is not really a bug. I saw this
http://lists.freeswitch.org/pipermail/freeswitch-users/2011-October/077379.html

In bypass_media mode also freeswitch tries to make sense of codec (pass
through codecs) a in my case g729a is receied, freeswitch does not
recognize it.

On Sat, Jun 11, 2016 at 9:37 PM, John Nash <john.nash778 at gmail.com> wrote:

> Ok will fill I was not sure it was a bug.
>
> On Fri, Jun 10, 2016 at 5:49 PM, Ken Rice <krice at freeswitch.org> wrote:
>
>> Buts should be filed at https://freeswitch.org/jira
>>
>>
>>
>> Please Attach a complete log of a call doing this (see the note at the
>> top of jira right under the menu bar on file attachments)
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *John Nash
>> *Sent:* Friday, June 10, 2016 1:33 AM
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* [Freeswitch-users] Freeswitch changes payload type of codec
>>
>>
>>
>> I am facing same issue described in this old post
>> http://lists.freeswitch.org/pipermail/freeswitch-users/2009-June/042823.html
>>
>>
>>
>> I am using media_bypass mode.
>>
>>
>>
>> If any more information like traces etc needed I can provide.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From ak at hejdu.dk  Fri Jun 17 17:16:47 2016
From: ak at hejdu.dk (Allan Kristensen)
Date: Fri, 17 Jun 2016 15:16:47 +0200
Subject: [Freeswitch-users] Does Freeswtich need rtcp attribute in sdp to
	enable rtcp?
Message-ID: <CAA4dF91ygkf6ZDMFO6Z3Se6gtpPa30LRnx5KU6=FYCMPz6Q_oA@mail.gmail.com>

Hello,

Does Freeswitch need the rtcp attribute in the sdp to enable rtcp?

When connecting to asterisk (which does not include a rtcp attribute),
Freeswitch does not start to listen on the rtcp port (rtp port +1).

When connecting to other Freeswitch servers, which have this attribute in
the sdp everything works fine.

According to the RFC 3605 it should not be needed, unless it doesn't follow
the parity rule (rtp port +1).

...Or is it just me?

Best regards,
  Allan
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From colton.conor at gmail.com  Fri Jun 17 17:24:29 2016
From: colton.conor at gmail.com (Colton Conor)
Date: Fri, 17 Jun 2016 08:24:29 -0500
Subject: [Freeswitch-users] 3GPP Enhanced Voice Services (EVS) codec
Message-ID: <CAMDdSzPP0tecDA2jKkPiww9fnJJ+J6k3eOwGDgkFfE5d1TZj_Q@mail.gmail.com>

Any chance that freeswitch will support the 3GPP Enhanced Voice Services
(EVS) codec? This codec seems to be that the codec LTE carriers are using
for enhance HD voice. According to this paper, this codec is even better
than OPUS:

http://networks.nokia.com/sites/default/files/document/nokia_3gpp_evs_codec_white_paper_0.pdf
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From mike at jerris.com  Fri Jun 17 17:44:19 2016
From: mike at jerris.com (Michael Jerris)
Date: Fri, 17 Jun 2016 09:44:19 -0400
Subject: [Freeswitch-users] Does Freeswtich need rtcp attribute in sdp
	to enable rtcp?
In-Reply-To: <CAA4dF91ygkf6ZDMFO6Z3Se6gtpPa30LRnx5KU6=FYCMPz6Q_oA@mail.gmail.com>
References: <CAA4dF91ygkf6ZDMFO6Z3Se6gtpPa30LRnx5KU6=FYCMPz6Q_oA@mail.gmail.com>
Message-ID: <B26B85E5-30CF-4BF0-8E21-D27B1CEE6B9D@jerris.com>

upgrade to latest release.

> On Jun 17, 2016, at 9:16 AM, Allan Kristensen <ak at hejdu.dk> wrote:
> 
> Hello,
> 
> Does Freeswitch need the rtcp attribute in the sdp to enable rtcp?
> 
> When connecting to asterisk (which does not include a rtcp attribute), Freeswitch does not start to listen on the rtcp port (rtp port +1).
> 
> When connecting to other Freeswitch servers, which have this attribute in the sdp everything works fine.
> 
> According to the RFC 3605 it should not be needed, unless it doesn't follow the parity rule (rtp port +1).
> 
> ...Or is it just me?
> 
> Best regards,
>   Allan
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Fri Jun 17 17:45:34 2016
From: mike at jerris.com (Michael Jerris)
Date: Fri, 17 Jun 2016 09:45:34 -0400
Subject: [Freeswitch-users] 3GPP Enhanced Voice Services (EVS) codec
In-Reply-To: <CAMDdSzPP0tecDA2jKkPiww9fnJJ+J6k3eOwGDgkFfE5d1TZj_Q@mail.gmail.com>
References: <CAMDdSzPP0tecDA2jKkPiww9fnJJ+J6k3eOwGDgkFfE5d1TZj_Q@mail.gmail.com>
Message-ID: <BA493A42-7CAE-45FE-94F6-673965C1E0C5@jerris.com>

As much chance as that codec not having a bunch of patents on it?

> On Jun 17, 2016, at 9:24 AM, Colton Conor <colton.conor at gmail.com> wrote:
> 
> Any chance that freeswitch will support the 3GPP Enhanced Voice Services (EVS) codec? This codec seems to be that the codec LTE carriers are using for enhance HD voice. According to this paper, this codec is even better than OPUS:
> 
> http://networks.nokia.com/sites/default/files/document/nokia_3gpp_evs_codec_white_paper_0.pdf <http://networks.nokia.com/sites/default/files/document/nokia_3gpp_evs_codec_white_paper_0.pdf>

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From daveh at beachdognet.com  Fri Jun 17 19:06:35 2016
From: daveh at beachdognet.com (Dave Horton)
Date: Fri, 17 Jun 2016 11:06:35 -0400
Subject: [Freeswitch-users] issue with uuid_set_media_stats showing zero
	packet counts
References: <20DD3566-4117-4099-ADE5-1C6DCA537B61@phone.com>
Message-ID: <20F79CF3-5AC3-497C-86DF-F0ECFAFE8E64@beachdognet.com>

I am trying to get media stats on my call, using the following dialplan:

  <extension name="announcement">
     <condition field="${sip_req_user}" expression="^annc$"/>
     <condition field="${sip_req_params}" expression="^play=(.*)$">
       <action application="answer"/>
       <action application="set" data="api_result=${uuid_set_media_stats(${uuid})}"/>
       <action application="sleep" data="1000"/>
       <action application="playback" data="http_cache://http://wcs-icr.s3.amazonaws.com/$1"/>
       <action application="set" data="api_result=${uuid_dump(${uuid})"/>
       <action application="log" data="incoming bytes ${rtp_audio_in_raw_bytes}"/>
       <action application="log" data="incoming bytes ${rtp_audio_out_raw_bytes}"/>
       <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/>
     </condition>
  </extension>

However, the data that is dumped shows packet counts of zero.  The prompt is playing, so we are certainly sending packets out.  

Any ideas?

Here is some of the output from the uuid_dump:

variable_endpoint_disposition: ANSWER
variable_rtp_audio_in_raw_bytes: 0
variable_rtp_audio_in_media_bytes: 0
variable_rtp_audio_in_packet_count: 0
variable_rtp_audio_in_media_packet_count: 0
variable_rtp_audio_in_skip_packet_count: 0
variable_rtp_audio_in_jitter_packet_count: 0
variable_rtp_audio_in_dtmf_packet_count: 0
variable_rtp_audio_in_cng_packet_count: 0
variable_rtp_audio_in_flush_packet_count: 0
variable_rtp_audio_in_largest_jb_size: 0
variable_rtp_audio_in_jitter_min_variance: 0.00
variable_rtp_audio_in_jitter_max_variance: 0.00
variable_rtp_audio_in_jitter_loss_rate: 0.00
variable_rtp_audio_in_jitter_burst_rate: 0.00
variable_rtp_audio_in_mean_interval: 0.00
variable_rtp_audio_in_flaw_total: 0
variable_rtp_audio_in_quality_percentage: 100.00
variable_rtp_audio_in_mos: 4.50
variable_rtp_audio_out_raw_bytes: 0
variable_rtp_audio_out_media_bytes: 0
variable_rtp_audio_out_packet_count: 0
variable_rtp_audio_out_media_packet_count: 0
variable_rtp_audio_out_skip_packet_count: 0
variable_rtp_audio_out_dtmf_packet_count: 0
variable_rtp_audio_out_cng_packet_count: 0
variable_rtp_audio_rtcp_packet_count: 0
variable_rtp_audio_rtcp_octet_count: 0

EXECUTE sofia/drachtio_mrf/5083084809 at 68.64.80.16 log(incoming bytes 0)
2016-06-17 10:59:28.757502 [DEBUG] mod_dptools.c:1713 bytes 0
EXECUTE sofia/drachtio_mrf/5083084809 at 68.64.80.16 log(incoming bytes 0)
2016-06-17 10:59:28.757502 [DEBUG] mod_dptools.c:1713 bytes 0





From bipin at xbipin.com  Fri Jun 17 20:03:00 2016
From: bipin at xbipin.com (Bipin Patel)
Date: Fri, 17 Jun 2016 20:03:00 +0400
Subject: [Freeswitch-users] any1 using opus in zoiper with FS successfully?
Message-ID: <229f2220-82ae-4a85-7613-ed762e9914a3@xbipin.com>

is there any1 using opus codec with FS using zoiper on android, coz im 
banging my head from past few days but calling from zoiper to zoiper 
always gives one way audio even though the other way RTP reaches FS but 
it doesnt send it back to zoiper and thats the cause of one way audio. I 
tried playing with the opus settings in almost all combinations but it 
just doesnt work, my profile has late negotiate and inherit codec 
enabled, tried disabling inherit codec but its still the same. FS cli 
also doesnt show any error regarding y it isnt sending the RTP back to 
dialer even though it gets it from the other zoiper

the packet trace is there on jira FS-9270 if any1 wants to investigate, 
any help is appreciated

-- 
Regards,
Bipin





From fvillarroel at yahoo.com  Fri Jun 17 20:17:13 2016
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Fri, 17 Jun 2016 16:17:13 +0000 (UTC)
Subject: [Freeswitch-users] popup web api
References: <772021148.5211361.1466180233082.JavaMail.yahoo.ref@mail.yahoo.com>
Message-ID: <772021148.5211361.1466180233082.JavaMail.yahoo@mail.yahoo.com>

Dear All.
I need your help in order to do the follow:
If i have a softphone registered on FS, so i need that when the peer (softphone) answered a inbound call in the browser display a form popup.
Somebody could help about how i cand do or what module i can use for do (mod_esl, mod_xml_curl....) or show me some example.
I appreciate some idea or example please.
Regards.

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From kevin.long at haloprivacy.com  Fri Jun 17 23:43:42 2016
From: kevin.long at haloprivacy.com (Kevin Long)
Date: Fri, 17 Jun 2016 19:43:42 +0000
Subject: [Freeswitch-users] Anyone doing SMS for their end users?
Message-ID: <87DA1211-AFD3-4C3E-BEE0-85E06EEB89FC@haloprivacy.com>


I am curious if anyone on this list is doing SMS for their SIP end users,  and if so,  how.  

Which SIP providers are you using,  how are you receiving and sending SMS to the outside world,  and what SIP or chat client software are you using to facilitate this?

Thank you,

Kevin Long





From schoch+freeswitch.org at xwin32.com  Sat Jun 18 00:40:12 2016
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Fri, 17 Jun 2016 13:40:12 -0700
Subject: [Freeswitch-users] Anyone doing SMS for their end users?
In-Reply-To: <87DA1211-AFD3-4C3E-BEE0-85E06EEB89FC@haloprivacy.com>
References: <87DA1211-AFD3-4C3E-BEE0-85E06EEB89FC@haloprivacy.com>
Message-ID: <CAOkuKdfAczPo5dtxuvBsyitr20kTQkay9TbsDSpazH1VG7Yc0g@mail.gmail.com>

I don't have many end users (using FreeSWITCH as our office PBX), but I've
just started using Flowroute's SMS API, and it works well.

I have possible future plans, but at the moment I have a PHP script that
takes any SMS message received on any of our DID's and turns it into an
email. Obviously the volume is low at the moment, so I just forward the
message to the recipient, but eventually I want to build an email to SMS
gateway.

Read my earlier message on this list for how I am using this SMS API to
send a text message when a voice mail is received, using a LUA script,
mod_curl, and the Flowroute SMS API.

-- 
Steve

On Fri, Jun 17, 2016 at 12:43 PM, Kevin Long <kevin.long at haloprivacy.com>
wrote:

>
> I am curious if anyone on this list is doing SMS for their SIP end users,
> and if so,  how.
>
> Which SIP providers are you using,  how are you receiving and sending SMS
> to the outside world,  and what SIP or chat client software are you using
> to facilitate this?
>
> Thank you,
>
> Kevin Long
>
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From naveen32india at gmail.com  Sat Jun 18 00:44:54 2016
From: naveen32india at gmail.com (Naveen Tamanam)
Date: Sat, 18 Jun 2016 02:14:54 +0530
Subject: [Freeswitch-users] CHANNEL_BRIDGE event is not coming when bridging
 two calls which are already in answered state
Message-ID: <CABsWbRYYdUsMqArtupX46kdZBXR2N42iacYUzCk=O3WGtpqK9A@mail.gmail.com>

Hi ,

CHANNEL_BRIDGE event is not coming when bridging two calls which are
already in answered state.
            Am I missing any thing? Why EVENT CHANNEL_BRIDGE is not coming
when bridging two calls which are in answered state.



-- 
Thanks & Regards,
Naveen Tamanam
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From blackc2004 at gmail.com  Sat Jun 18 01:38:47 2016
From: blackc2004 at gmail.com (Cj B)
Date: Fri, 17 Jun 2016 14:38:47 -0700
Subject: [Freeswitch-users] Anyone doing SMS for their end users?
In-Reply-To: <87DA1211-AFD3-4C3E-BEE0-85E06EEB89FC@haloprivacy.com>
References: <87DA1211-AFD3-4C3E-BEE0-85E06EEB89FC@haloprivacy.com>
Message-ID: <CAAKko=GvQn3fUwbUUnw6s5sUBXJ1CXtH=Fp6mhGi0skuG=1N4g@mail.gmail.com>

I wrote an add-on app to FusionPBX (
https://github.com/fusionpbx/fusionpbx-apps/tree/master/sms)  which works
with flowroute, twilio and teli. It sends/receives the message and forwards
it to the end point (tested with yealink phones). I also just recently
submitted code for voicemail to sms with the same add-on. Wouldn't be hard
to extend it to do sms to email, etc.

So far it's working pretty good for me but it's still in limited use in my
production environment.

On Fri, Jun 17, 2016 at 12:43 PM, Kevin Long <kevin.long at haloprivacy.com>
wrote:

>
> I am curious if anyone on this list is doing SMS for their SIP end users,
> and if so,  how.
>
> Which SIP providers are you using,  how are you receiving and sending SMS
> to the outside world,  and what SIP or chat client software are you using
> to facilitate this?
>
> Thank you,
>
> Kevin Long
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From max at nysolutions.com  Sat Jun 18 01:44:49 2016
From: max at nysolutions.com (Moishe Grunstein)
Date: Fri, 17 Jun 2016 21:44:49 +0000
Subject: [Freeswitch-users] Anyone doing SMS for their end users?
In-Reply-To: <87DA1211-AFD3-4C3E-BEE0-85E06EEB89FC@haloprivacy.com>
References: <87DA1211-AFD3-4C3E-BEE0-85E06EEB89FC@haloprivacy.com>
Message-ID: <e2c6386c37b74b3bacf1629d9c24eacf@nysolutions.com>

We use playsms

Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US
Service Request Email: support at nysolutions.com 

Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kevin Long
Sent: Friday, June 17, 2016 3:44 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Anyone doing SMS for their end users?


I am curious if anyone on this list is doing SMS for their SIP end users,  and if so,  how.  

Which SIP providers are you using,  how are you receiving and sending SMS to the outside world,  and what SIP or chat client software are you using to facilitate this?

Thank you,

Kevin Long




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

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

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


From ssinyagin at gmail.com  Sat Jun 18 05:44:04 2016
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sat, 18 Jun 2016 03:44:04 +0200
Subject: [Freeswitch-users] Anyone doing SMS for their end users?
In-Reply-To: <87DA1211-AFD3-4C3E-BEE0-85E06EEB89FC@haloprivacy.com>
References: <87DA1211-AFD3-4C3E-BEE0-85E06EEB89FC@haloprivacy.com>
Message-ID: <CACOYK=ihdrHfeNShTUA9xPdMQCWxATBiEz1AWGKBNeX5yV3-uQ@mail.gmail.com>

there are only few countries where you can buy a virtual number that
is able to receive both SMS and voice calls. So, we ended up with a
bunch of Yeastar GSM gateways with SIM cards in them. The gateways are
quite stable, and the only issue is that if an SMS starts with a
smiley, you get an empty message.

We're receiving the SMS from those gateways via their Asterisk API,
and insert them into a Redis queue. Then, there's a dispatcher that
reads the queue and forwards the messages depending on the user's
follow-me status: it can be sent to a Telegram messenger, or as a new
SMS to the user's mobile phone.  Then there's a web app where the user
can see the messages and reply to them.

We use http://www.inetworx.ch/ for outbound SMS. They allow setting
the sender's number, and that is very handy in this scenario.

We use a local copy of Notifica bot for sending messages to Telegram:
https://storebot.me/bot/notifica_bot  The author is going to publish
it as open-source, but hasn't yet found the time for it:
https://github.com/Benez/notifica_bot


cheers,
stanislav


On Fri, Jun 17, 2016 at 9:43 PM, Kevin Long <kevin.long at haloprivacy.com> wrote:
>
> I am curious if anyone on this list is doing SMS for their SIP end users,  and if so,  how.
>
> Which SIP providers are you using,  how are you receiving and sending SMS to the outside world,  and what SIP or chat client software are you using to facilitate this?
>
> Thank you,
>
> Kevin Long
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From krice at freeswitch.org  Sat Jun 18 06:26:17 2016
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 17 Jun 2016 21:26:17 -0500
Subject: [Freeswitch-users] Anyone doing SMS for their end users?
In-Reply-To: <87DA1211-AFD3-4C3E-BEE0-85E06EEB89FC@haloprivacy.com>
References: <87DA1211-AFD3-4C3E-BEE0-85E06EEB89FC@haloprivacy.com>
Message-ID: <98075304-D790-4314-ABD9-46C08F679E29@freeswitch.org>

In the US many of the Wholesale DID providers are sms enabled these days companies like Flowroute and others have apis for handling this in the US, in the UK i have used Ziron

Sent from my iPhone

> On Jun 17, 2016, at 2:43 PM, Kevin Long <kevin.long at haloprivacy.com> wrote:
> 
> 
> I am curious if anyone on this list is doing SMS for their SIP end users,  and if so,  how.  
> 
> Which SIP providers are you using,  how are you receiving and sending SMS to the outside world,  and what SIP or chat client software are you using to facilitate this?
> 
> Thank you,
> 
> Kevin Long
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From richard.mace at gmail.com  Sat Jun 18 11:03:51 2016
From: richard.mace at gmail.com (Richard Mace)
Date: Sat, 18 Jun 2016 08:03:51 +0100
Subject: [Freeswitch-users] ANN RoCKSwitch 0.993 with multiple inbound
	destinations
In-Reply-To: <CAK+KT3_aNjt2Pu2kicdn_e8yC1ciguiQHKGUhANvcanbSXLv+A@mail.gmail.com>
References: <CAK+KT38QNC2aGGAFQcxj3yu8s9_J1G-z-w0funW1-qFiUbn_RA@mail.gmail.com>
	<CAK+KT3_aNjt2Pu2kicdn_e8yC1ciguiQHKGUhANvcanbSXLv+A@mail.gmail.com>
Message-ID: <CAK+KT39X=XtT=MKcq2BENqHZFBcr09=boyYnb4sHJZwvJyyJsg@mail.gmail.com>

Hi all,

Just a quick note to say that a new version of RoCKSwitch is available that
now supports multiple inbound destinations.

Would be really grateful for any feedback.

Richard
www.rocksoftware.co.uk
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From ashwinrkjain at gmail.com  Sat Jun 18 11:22:05 2016
From: ashwinrkjain at gmail.com (Ashwin Jain)
Date: Sat, 18 Jun 2016 12:52:05 +0530
Subject: [Freeswitch-users] Audio skip issue with simultaneous ringing
Message-ID: <CAM9SNVTQJo5jCZ+jb9Yy6fErb7UdUtH1QpbNn=8kPkdaP=m0vg@mail.gmail.com>

Hi all,

We are running Freeswitch version 1.4.26. We are facing a major problem
related to audio skip with simultaneous ringing. In one flow we are ringing
around 5 agents at the same time. Whenever agent 1 (any one of the agent)
picks up the call, other legs (to other agents) gets a hangup event. Due to
this, hangup event to all the other agent "CALL_ANSWER" event on the other
side gets delayed by around 2-3 seconds and generally caller missed the
first word from the agent.

Event 1: Call Answered by agent 1
Event 2: Hangup for agent 2 (due to LOSE_RACE)
Event 3: Hangup for agent 3 (due to LOSE_RACE)
Event 4: Hangup for agent 4 (due to LOSE_RACE)
Event 5: Hangup for agent 5 (due to LOSE_RACE)
Event 6: Call answered to caller

Time different between Event 1 and Event 6 is about 2-3 seconds.
Any help would be really really helpful

-- 
Thanks and Regards,
Ashwin Jain
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From krc at retrospekt.dk  Sat Jun 18 12:17:26 2016
From: krc at retrospekt.dk (Kim Rostgaard Christensen)
Date: Sat, 18 Jun 2016 10:17:26 +0200
Subject: [Freeswitch-users] popup web api
In-Reply-To: <772021148.5211361.1466180233082.JavaMail.yahoo@mail.yahoo.com>
References: <772021148.5211361.1466180233082.JavaMail.yahoo.ref@mail.yahoo.com>
	<772021148.5211361.1466180233082.JavaMail.yahoo@mail.yahoo.com>
Message-ID: <57650396.4030700@retrospekt.dk>

Hi

We are doing almost exactly this in our reception-frontdesk software 
stack[1]. It has a server backend that connects to ESL and wraps and 
forwards relevant event information via a websocket (or long polling) to 
clients.
To accomplish this, we use (my) ESL library. You can find an example 
client here: 
https://github.com/rostgaard/libesl-dart/blob/master/example/basic.dart
It is written in Dart, which has proven (for us) to be _very_ stable for 
daily use, at ~3k calls pr day.

Websocket server and client examples can be found here: 
https://www.dartlang.org/samples/websockets

Well, enough shameless plug for one day.

[1] https://github.com/Bitstackers/openreception

On 2016-06-17 18:17, FERNANDO VILLARROEL wrote:
> Dear All.
>
> I need your help in order to do the follow:
>
> If i have a softphone registered on FS, so i need that when the peer 
> (softphone) answered a inbound call in the browser display a form popup.
>
> Somebody could help about how i cand do or what module i can use for 
> do (mod_esl, mod_xml_curl....) or show me some example.
>
> I appreciate some idea or example please.
>
> Regards.
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From murat.knecht at googlemail.com  Sat Jun 18 12:43:15 2016
From: murat.knecht at googlemail.com (Murat Knecht)
Date: Sat, 18 Jun 2016 16:43:15 +0800
Subject: [Freeswitch-users] Anyone doing SMS for their end users?
In-Reply-To: <87DA1211-AFD3-4C3E-BEE0-85E06EEB89FC@haloprivacy.com>
References: <87DA1211-AFD3-4C3E-BEE0-85E06EEB89FC@haloprivacy.com>
Message-ID: <576509A3.5010808@googlemail.com>

Hey Kevin,

we are.

Assuming you're interested in blasting SMS out via API, there's a ton of
providers out there for that. I think the biggest, in the US at least,
should be Twilio [2]
There are others, that are cheaper, though.

For more complex scenarios, providers like TextIt [3] should be able to
help you. They allow you to create complex flows.

If you'd like a  web-UI to help you reach many people in complex
scenarios (polls, surveys, timing based curriculums, keyword-response
auto-replies), we at engageSPARK [1] might be able to help you.

Hope that gives you an idea of the spectrum, so you find the right tool
for your job.

Cheers,
murat


[1] https://www.engagespark.com/
[2] https://www.twilio.com/docs/api/rest/sending-messages
[3] https://textit.in/


On 06/18/2016 03:43 AM, Kevin Long wrote:
> I am curious if anyone on this list is doing SMS for their SIP end users,  and if so,  how.  
>
> Which SIP providers are you using,  how are you receiving and sending SMS to the outside world,  and what SIP or chat client software are you using to facilitate this?
>
> Thank you,
>
> Kevin Long
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From tanim05 at gmail.com  Sat Jun 18 13:31:46 2016
From: tanim05 at gmail.com (Tanim)
Date: Sat, 18 Jun 2016 15:31:46 +0600
Subject: [Freeswitch-users] How to playback background call process
In-Reply-To: <CACOYK=jaqMHsijwMjJTpUA-tcnCK+22822DUo5evgco6hZikyw@mail.gmail.com>
References: <000701d1c64f$65ff6f10$31fe4d30$@gmail.com>
	<CACJ9VGNY-hU59+S2mFzGeLO6YhHg7c53bjite6JTZXRQe3Bd9Q@mail.gmail.com>
	<00b801d1c715$76ef7420$64ce5c60$@gmail.com>
	<CACOYK=jaqMHsijwMjJTpUA-tcnCK+22822DUo5evgco6hZikyw@mail.gmail.com>
Message-ID: <008a01d1c944$3da66850$b8f338f0$@gmail.com>

I was tested with freeswitch 1.6.8

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stanislav Sinyagin
Sent: Friday, June 17, 2016 12:14 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How to playback background call process

 

You need to reproduce it with latest FreeSWITCH, and file a jira if the problem persists.

On Jun 15, 2016 16:53, "Tanim" <tanim05 at gmail.com> wrote:

Thanks for your reply. Its worked partially.

When FS get ringback from far end then its working fine. But if a call take more time to get RBT then FS gets crashed. Here is the log:

 

[DEBUG] switch_ivr_originate.c:1305 Play Ringback File [file_string://digits/youhave.wav!digits/9.wav]

[DEBUG] switch_rtp.c:1846 rtcp_stats_init: audio ssrc[37546101] base_seq[0]

[DEBUG] switch_rtp.c:6707 Correct audio ip/port confirmed.

freeswitch: src/switch_core_file.c:378: switch_core_file_read: Assertion `fh->file_interface != ((void *)0)' failed.

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of ??????? ???????
Sent: Wednesday, June 15, 2016 6:51 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How to playback background call process

 

Hi

Try to use <action application="set" data="ringback=/IVR/FILE" />

 

 

2016-06-14 18:14 GMT+03:00 Tanim <tanim05 at gmail.com>:

Hi,

 

I want to playback file background call process. So that in playing IVR time, call will process to send to gateways. Currently my dialplan like this:

<action application="playback" data="/IVR/FILE" />

<action application="bridge" data="sofia/gateway/gw1/123456" />

 

In this way before call processing IVR is playing then start to process the call. But I want both in same time.

Can you please help me, how can I do it.

 

Thanks in advance.

Tanim

 


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

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

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




-- 

WBR

Alex 


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

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

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

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From vfclists at gmail.com  Sat Jun 18 14:10:57 2016
From: vfclists at gmail.com (vfclists)
Date: Sat, 18 Jun 2016 11:10:57 +0100
Subject: [Freeswitch-users] What field type supports microsecond epochs in
	SQLite CDR?
Message-ID: <CAE3mkNB+x3d4y-K7zfNQkVDwTVXwLsRzahUtK82irz8J=+=Qpw@mail.gmail.com>

What field type in SQLite should be used to store the micro second epoch
fields like answer_uepoch. Should they be bigint or double?
-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From steveayre at gmail.com  Sun Jun 19 02:25:31 2016
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 18 Jun 2016 23:25:31 +0100
Subject: [Freeswitch-users] What field type supports microsecond epochs
 in SQLite CDR?
In-Reply-To: <CAE3mkNB+x3d4y-K7zfNQkVDwTVXwLsRzahUtK82irz8J=+=Qpw@mail.gmail.com>
References: <CAE3mkNB+x3d4y-K7zfNQkVDwTVXwLsRzahUtK82irz8J=+=Qpw@mail.gmail.com>
Message-ID: <CAFiqYunt4RRm3rEEKxw6k_bsTQuJVf-QDu85UTo+K2a6VZNNtw@mail.gmail.com>

Use bigint.

_uepoch values are always an integer not floating point. In sqlite bigint
and double are both (up to) 8 bytes so both have roughly the same range.



On 18 June 2016 at 11:10, vfclists <vfclists at gmail.com> wrote:

>
> What field type in SQLite should be used to store the micro second epoch
> fields like answer_uepoch. Should they be bigint or double?
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From joel at gogii.net  Sun Jun 19 04:27:46 2016
From: joel at gogii.net (Joel Serrano)
Date: Sat, 18 Jun 2016 17:27:46 -0700
Subject: [Freeswitch-users] calling all native speakers of languages
 other than english. We need your help, and this will be easy.
In-Reply-To: <f11ee79d-327f-c6d1-d8a8-dfede83d9a8f@middleearth.sapphiresunday.org>
References: <391E7162-7420-4CF9-A229-936DC3A5636D@jerris.com>
	<f11ee79d-327f-c6d1-d8a8-dfede83d9a8f@middleearth.sapphiresunday.org>
Message-ID: <CAMtXxQ=gvRC8kC6TYXtF-2dEQWWYMf7uHCrmL7Coub_Y9+2EpQ@mail.gmail.com>

Hi,

I'm translating this to Spanish, but certain phrases are very "context
dependent" so just a literal translation will be probably incorrect.

Before I continue, two questions:

1- Is there a DEMO of Verto I can login to, so I can see more or less where
each phrase is?
2- Has anyone else already done the translation for Spanish? I don't want
to step on anyone's work or put time into something that is already done.

Thanks!
Joel.

On Mon, Jun 6, 2016 at 2:50 PM, Trever L. Adams <
trever at middleearth.sapphiresunday.org> wrote:

> On 03/29/2016 02:15 PM, Michael Jerris wrote:
> > Just committed into tree is internationalization of Verto
> > Communicator, so far, just English and Italian.  If you speak another
> > language, we need your help to translate these into other languages.
> >  The file that needs to be translated is:
> >
> >
> https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/html5/verto/verto_communicator/src/locales/locale-en.json?at=6c197ae2f0cb647ab20bc5143870a80c3dd8d602&raw
> >
> > If you can assist with this, Please make a new file named
> > locale-language.json (where "language" is the 2 letter code for your
> > language) and translate all the text in quotes to the right thing for
> > your language and send them back our way.  Lets see how quickly we can
> > crowd source this
> >
> > Thanks!
> I am not a native speaker of Hebrew, but I have been learning for
> several years and can read, write and speak like a child (and more
> complex documents such as the Tanakh). I cannot say this is error free,
> but it is MUCH better than what a machine translation would likely be. I
> am not seeing these files in my local repository for some reason.
>
> If you are interested, here it is.
>
> Thank you,
> Trever
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ashwinrkjain at gmail.com  Sun Jun 19 19:36:59 2016
From: ashwinrkjain at gmail.com (Ashwin Jain)
Date: Sun, 19 Jun 2016 21:06:59 +0530
Subject: [Freeswitch-users] Audio skip issue with simultaneous ringing
In-Reply-To: <CAM9SNVTQJo5jCZ+jb9Yy6fErb7UdUtH1QpbNn=8kPkdaP=m0vg@mail.gmail.com>
References: <CAM9SNVTQJo5jCZ+jb9Yy6fErb7UdUtH1QpbNn=8kPkdaP=m0vg@mail.gmail.com>
Message-ID: <CAM9SNVRpSMnm+fdF9Zq9haZMf3SYtQW99q2HxM0eEzz6wHQbdA@mail.gmail.com>

Any input on this would be really helpful.

On Sat, Jun 18, 2016 at 12:52 PM, Ashwin Jain <ashwinrkjain at gmail.com>
wrote:

> Hi all,
>
> We are running Freeswitch version 1.4.26. We are facing a major problem
> related to audio skip with simultaneous ringing. In one flow we are ringing
> around 5 agents at the same time. Whenever agent 1 (any one of the agent)
> picks up the call, other legs (to other agents) gets a hangup event. Due to
> this, hangup event to all the other agent "CALL_ANSWER" event on the other
> side gets delayed by around 2-3 seconds and generally caller missed the
> first word from the agent.
>
> Event 1: Call Answered by agent 1
> Event 2: Hangup for agent 2 (due to LOSE_RACE)
> Event 3: Hangup for agent 3 (due to LOSE_RACE)
> Event 4: Hangup for agent 4 (due to LOSE_RACE)
> Event 5: Hangup for agent 5 (due to LOSE_RACE)
> Event 6: Call answered to caller
>
> Time different between Event 1 and Event 6 is about 2-3 seconds.
> Any help would be really really helpful
>
> --
> Thanks and Regards,
> Ashwin Jain
>



-- 

*Ashwin Jain*
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From vfclists at gmail.com  Mon Jun 20 05:41:29 2016
From: vfclists at gmail.com (vfclists)
Date: Mon, 20 Jun 2016 02:41:29 +0100
Subject: [Freeswitch-users] Timestamp and timezone issues in CDR,
	bug or by design?
Message-ID: <CAE3mkNAwnmjRoYNtC22Na1QbzSf3xixy6e1eB0XBjpeZvNVFgA@mail.gmail.com>

When I check the relationship between the epochs and the timestamps, eg
answer_epoch vs answer_stamp I realize that after the daylight saving
change the epochs are an hour behind. Is this by design? Are the timestamps
supposed to be based on the computer's timezone by design, with the epochs
based on the UTC value regardless of timezone and daylight savings?

select
(julianday('2016-03-30 15:23:00')-2440587.5) * 86400.0, --
1459351379.999998, answer_epoch is 1459347780
 (julianday('2016-03-30 14:23:00')-2440587.5) * 86400.0, --
1459347780.000012, value above - 1hr
(julianday('2016-01-27 11:58:53')-2440587.5) * 86400 --1453895933.000001 -
prior to daylight savings

The first 2 are from the sqlite CDR and the last from the CSV file.



-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From nneul at mst.edu  Mon Jun 20 06:38:45 2016
From: nneul at mst.edu (Nathan Neulinger)
Date: Sun, 19 Jun 2016 21:38:45 -0500
Subject: [Freeswitch-users] Timestamp and timezone issues in CDR,
 bug or by design?
In-Reply-To: <CAE3mkNAwnmjRoYNtC22Na1QbzSf3xixy6e1eB0XBjpeZvNVFgA@mail.gmail.com>
References: <CAE3mkNAwnmjRoYNtC22Na1QbzSf3xixy6e1eB0XBjpeZvNVFgA@mail.gmail.com>
Message-ID: <80b8be5f-f941-626a-557e-9f2abbc03152@mst.edu>

epoch type timestamps are always UTC by definition, it's a universal/exact format.

-- Nathan

On 06/19/2016 08:41 PM, vfclists wrote:
>
> When I check the relationship between the epochs and the timestamps, eg answer_epoch vs answer_stamp I realize that
> after the daylight saving change the epochs are an hour behind. Is this by design? Are the timestamps supposed to be
> based on the computer's timezone by design, with the epochs based on the UTC value regardless of timezone and daylight
> savings?
>
> select
> (julianday('2016-03-30 15:23:00')-2440587.5) * 86400.0, -- 1459351379.999998, answer_epoch is 1459347780
>  (julianday('2016-03-30 14:23:00')-2440587.5) * 86400.0, -- 1459347780.000012, value above - 1hr
> (julianday('2016-01-27 11:58:53')-2440587.5) * 86400 --1453895933.000001 - prior to daylight savings
>
> The first 2 are from the sqlite CDR and the last from the CSV file.
>
>
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.com
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

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


From 568691 at gmail.com  Mon Jun 20 06:58:06 2016
From: 568691 at gmail.com (Alexandru Covalschi)
Date: Mon, 20 Jun 2016 05:58:06 +0300
Subject: [Freeswitch-users] Tone detection issue
Message-ID: <CAD_iGr-5T60b0RbYb0n5A53KsQak7sDwtnSkt0Jqxf8WmnBr5Q@mail.gmail.com>

Hi list,

I have an issue using tone_detect. The thing is I must detect two very
short tones - 1400 Hz and 2300 Hz 100ms coming one after another (with 100
ms pause I suppose) after ~2 seconds the call is answered. I have pretty
accurate wav file, in audacity I see pikes are 1400 and 2296.
I've seen that in spandsp sources it is said that accuracy is ~3% so that
should be enough,
However tones aren't detected even if I set:

<action application="tone_detect" data="tone_1_0 1395 r 0 set
tone_first=true 1"/>
<action application="tone_detect" data="tone_2_0 2295 r 0 set
tone_second=true 1"/>
<action application="tone_detect" data="tone_1_1 1396 r 0 set
tone_first=true 1"/>
<action application="tone_detect" data="tone_2_1 2296 r 0 set
tone_second=true 1"/>
<action application="tone_detect" data="tone_1_2 1397 r 0 set
tone_first=true 1"/>
<action application="tone_detect" data="tone_2_2 2297 r 0 set
tone_second=true 1"/>
<action application="tone_detect" data="tone_1_3 1398 r 0 set
tone_first=true 1"/>
<action application="tone_detect" data="tone_2_3 2298 r 0 set
tone_second=true 1"/>
<action application="tone_detect" data="tone_1_4 1399 r 0 set
tone_first=true 1"/>
<action application="tone_detect" data="tone_2_4 2299 r 0 set
tone_second=true 1"/>
<action application="tone_detect" data="tone_1_5 1400 r 0 set
tone_first=true 1"/>
<action application="tone_detect" data="tone_2_5 2300 r 0 set
tone_second=true 1"/>
<action application="tone_detect" data="tone_1_6 1401 r 0 set
tone_first=true 1"/>
<action application="tone_detect" data="tone_2_6 2301 r 0 set
tone_second=true 1"/>
<action application="tone_detect" data="tone_1_7 1402 r 0 set
tone_first=true 1"/>
<action application="tone_detect" data="tone_2_7 2302 r 0 set
tone_second=true 1"/>

Tried to do same using spandsp_start_tone_detect(), but I have no idea how
to edit spandsp.conf to add a tone matching my needs.

Can someone give me any clue? Any parameter I need to change in sources to
obtain what I need?

Debian Jessie
FreeSWITCH version: 1.6.6-13-d2d0b32~64bit (-13-d2d0b32 64bit)


Thanks in advance!
-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
tel: +37367398493
web: http://abriss.solutions/ <http://abs-telecom.com/>
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From mike at jerris.com  Mon Jun 20 13:46:18 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 20 Jun 2016 05:46:18 -0400
Subject: [Freeswitch-users] How to playback background call process
In-Reply-To: <008a01d1c944$3da66850$b8f338f0$@gmail.com>
References: <000701d1c64f$65ff6f10$31fe4d30$@gmail.com>
	<CACJ9VGNY-hU59+S2mFzGeLO6YhHg7c53bjite6JTZXRQe3Bd9Q@mail.gmail.com>
	<00b801d1c715$76ef7420$64ce5c60$@gmail.com>
	<CACOYK=jaqMHsijwMjJTpUA-tcnCK+22822DUo5evgco6hZikyw@mail.gmail.com>
	<008a01d1c944$3da66850$b8f338f0$@gmail.com>
Message-ID: <CAEwTsAi7jXcQGL-BggxLAcGd6+50B1ZBW19NUCi_W1ZDGoBG0w@mail.gmail.com>

You need to reproduce it with latest FreeSWITCH, and file a jira if the
problem persists.

On Saturday, June 18, 2016, Tanim <tanim05 at gmail.com> wrote:

> I was tested with freeswitch 1.6.8
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>]
> *On Behalf Of *Stanislav Sinyagin
> *Sent:* Friday, June 17, 2016 12:14 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] How to playback background call process
>
>
>
> You need to reproduce it with latest FreeSWITCH, and file a jira if the
> problem persists.
>
> On Jun 15, 2016 16:53, "Tanim" <tanim05 at gmail.com
> <javascript:_e(%7B%7D,'cvml','tanim05 at gmail.com');>> wrote:
>
> Thanks for your reply. Its worked partially.
>
> When FS get ringback from far end then its working fine. But if a call
> take more time to get RBT then FS gets crashed. Here is the log:
>
>
>
> [DEBUG] switch_ivr_originate.c:1305 Play Ringback File
> [file_string://digits/youhave.wav!digits/9.wav]
>
> [DEBUG] switch_rtp.c:1846 rtcp_stats_init: audio ssrc[37546101] base_seq[0]
>
> [DEBUG] switch_rtp.c:6707 Correct audio ip/port confirmed.
>
> freeswitch: src/switch_core_file.c:378: switch_core_file_read: Assertion
> `fh->file_interface != ((void *)0)' failed.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>]
> *On Behalf Of *??????? ???????
> *Sent:* Wednesday, June 15, 2016 6:51 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] How to playback background call process
>
>
>
> Hi
>
> Try to use <action application="set" data="ringback=/IVR/FILE" />
>
>
>
>
>
> 2016-06-14 18:14 GMT+03:00 Tanim <tanim05 at gmail.com
> <javascript:_e(%7B%7D,'cvml','tanim05 at gmail.com');>>:
>
> Hi,
>
>
>
> I want to playback file background call process. So that in playing IVR
> time, call will process to send to gateways. Currently my dialplan like
> this:
>
> <action application="playback" data="/IVR/FILE" />
>
> <action application="bridge" data="sofia/gateway/gw1/123456" />
>
>
>
> In this way before call processing IVR is playing then start to process
> the call. But I want both in same time.
>
> Can you please help me, how can I do it.
>
>
>
> Thanks in advance.
>
> Tanim
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
>
> WBR
>
> Alex
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Mon Jun 20 13:49:56 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 20 Jun 2016 05:49:56 -0400
Subject: [Freeswitch-users] calling all native speakers of languages
 other than english. We need your help, and this will be easy.
In-Reply-To: <CAMtXxQ=gvRC8kC6TYXtF-2dEQWWYMf7uHCrmL7Coub_Y9+2EpQ@mail.gmail.com>
References: <391E7162-7420-4CF9-A229-936DC3A5636D@jerris.com>
	<f11ee79d-327f-c6d1-d8a8-dfede83d9a8f@middleearth.sapphiresunday.org>
	<CAMtXxQ=gvRC8kC6TYXtF-2dEQWWYMf7uHCrmL7Coub_Y9+2EpQ@mail.gmail.com>
Message-ID: <CAEwTsAjS8SHSDw9QNmTs=zC_75_PZD=rsfyse5MyQopvmK4j7Q@mail.gmail.com>

read back in this thread around the end of March

On Saturday, June 18, 2016, Joel Serrano <joel at gogii.net> wrote:

> Hi,
>
> I'm translating this to Spanish, but certain phrases are very "context
> dependent" so just a literal translation will be probably incorrect.
>
> Before I continue, two questions:
>
> 1- Is there a DEMO of Verto I can login to, so I can see more or less
> where each phrase is?
> 2- Has anyone else already done the translation for Spanish? I don't want
> to step on anyone's work or put time into something that is already done.
>
> Thanks!
> Joel.
>
> On Mon, Jun 6, 2016 at 2:50 PM, Trever L. Adams <
> trever at middleearth.sapphiresunday.org
> <javascript:_e(%7B%7D,'cvml','trever at middleearth.sapphiresunday.org');>>
> wrote:
>
>> On 03/29/2016 02:15 PM, Michael Jerris wrote:
>> > Just committed into tree is internationalization of Verto
>> > Communicator, so far, just English and Italian.  If you speak another
>> > language, we need your help to translate these into other languages.
>> >  The file that needs to be translated is:
>> >
>> >
>> https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/html5/verto/verto_communicator/src/locales/locale-en.json?at=6c197ae2f0cb647ab20bc5143870a80c3dd8d602&raw
>> >
>> > If you can assist with this, Please make a new file named
>> > locale-language.json (where "language" is the 2 letter code for your
>> > language) and translate all the text in quotes to the right thing for
>> > your language and send them back our way.  Lets see how quickly we can
>> > crowd source this
>> >
>> > Thanks!
>> I am not a native speaker of Hebrew, but I have been learning for
>> several years and can read, write and speak like a child (and more
>> complex documents such as the Tanakh). I cannot say this is error free,
>> but it is MUCH better than what a machine translation would likely be. I
>> am not seeing these files in my local repository for some reason.
>>
>> If you are interested, here it is.
>>
>> Thank you,
>> Trever
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From lists at kavun.ch  Mon Jun 20 14:29:33 2016
From: lists at kavun.ch (Emrah)
Date: Mon, 20 Jun 2016 12:29:33 +0200
Subject: [Freeswitch-users] Audio skip issue with simultaneous ringing
In-Reply-To: <CAM9SNVRpSMnm+fdF9Zq9haZMf3SYtQW99q2HxM0eEzz6wHQbdA@mail.gmail.com>
References: <CAM9SNVTQJo5jCZ+jb9Yy6fErb7UdUtH1QpbNn=8kPkdaP=m0vg@mail.gmail.com>
	<CAM9SNVRpSMnm+fdF9Zq9haZMf3SYtQW99q2HxM0eEzz6wHQbdA@mail.gmail.com>
Message-ID: <8F5AA30D-C8FD-48BA-91B0-141FCCA286BF@kavun.ch>

Hello,
Can you share your bridge string as well as some logs?
What kind of processor are you running this on?
Did you source compile your FS or are you using a packaged version?

Emrah
> On Jun 19, 2016, at 5:36 PM, Ashwin Jain <ashwinrkjain at gmail.com> wrote:
> 
> Any input on this would be really helpful.
> 
> On Sat, Jun 18, 2016 at 12:52 PM, Ashwin Jain <ashwinrkjain at gmail.com <mailto:ashwinrkjain at gmail.com>> wrote:
> Hi all,
> 
> We are running Freeswitch version 1.4.26. We are facing a major problem related to audio skip with simultaneous ringing. In one flow we are ringing around 5 agents at the same time. Whenever agent 1 (any one of the agent) picks up the call, other legs (to other agents) gets a hangup event. Due to this, hangup event to all the other agent "CALL_ANSWER" event on the other side gets delayed by around 2-3 seconds and generally caller missed the first word from the agent.
> 
> Event 1: Call Answered by agent 1
> Event 2: Hangup for agent 2 (due to LOSE_RACE)
> Event 3: Hangup for agent 3 (due to LOSE_RACE)
> Event 4: Hangup for agent 4 (due to LOSE_RACE)
> Event 5: Hangup for agent 5 (due to LOSE_RACE)
> Event 6: Call answered to caller
> 
> Time different between Event 1 and Event 6 is about 2-3 seconds.
> Any help would be really really helpful
> 
> -- 
> Thanks and Regards,
> Ashwin Jain 
> 
> 
> 
> -- 
> Ashwin Jain
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From vfclists at gmail.com  Mon Jun 20 15:12:34 2016
From: vfclists at gmail.com (vfclists)
Date: Mon, 20 Jun 2016 12:12:34 +0100
Subject: [Freeswitch-users] Timestamp and timezone issues in CDR,
 bug or by design?
In-Reply-To: <80b8be5f-f941-626a-557e-9f2abbc03152@mst.edu>
References: <CAE3mkNAwnmjRoYNtC22Na1QbzSf3xixy6e1eB0XBjpeZvNVFgA@mail.gmail.com>
	<80b8be5f-f941-626a-557e-9f2abbc03152@mst.edu>
Message-ID: <CAE3mkNDWZ8fZuPCXFB=rQWYHFqyt7QES8NLOw6ah7P1_NhX+4Q@mail.gmail.com>

It is what I was thinking. I just wanted to be sure.

Do service providers generally present the CDRs using UTC time or do they
adjust them to the customers time zone?

On 20 June 2016 at 03:38, Nathan Neulinger <nneul at mst.edu> wrote:

> epoch type timestamps are always UTC by definition, it's a universal/exact
> format.
>
> -- Nathan
>
> On 06/19/2016 08:41 PM, vfclists wrote:
> >
> > When I check the relationship between the epochs and the timestamps, eg
> answer_epoch vs answer_stamp I realize that
> > after the daylight saving change the epochs are an hour behind. Is this
> by design? Are the timestamps supposed to be
> > based on the computer's timezone by design, with the epochs based on the
> UTC value regardless of timezone and daylight
> > savings?
> >
> > select
> > (julianday('2016-03-30 15:23:00')-2440587.5) * 86400.0, --
> 1459351379.999998, answer_epoch is 1459347780
> >  (julianday('2016-03-30 14:23:00')-2440587.5) * 86400.0, --
> 1459347780.000012, value above - 1hr
> > (julianday('2016-01-27 11:58:53')-2440587.5) * 86400 --1453895933.000001
> - prior to daylight savings
> >
> > The first 2 are from the sqlite CDR and the last from the CSV file.
> >
> >
> >
> > --
> > Frank Church
> http://devblog.brahmancreations.com
>
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From aubalde at presenceco.com  Mon Jun 20 15:14:21 2016
From: aubalde at presenceco.com (=?UTF-8?Q?Agust=C3=AD_Ubalde?=)
Date: Mon, 20 Jun 2016 13:14:21 +0200
Subject: [Freeswitch-users] 2 calls to the same extension
Message-ID: <CAKThY07NVaJmcOd=CtghiihFnBW=jsxbkUPF+8tTCS2nawVJrg@mail.gmail.com>

Hi all,

Would be possible to bridge 2 calls about the same local extension? Or is
it better to use a conference?


Thanks!

*PRESENCE TECHNOLOGY*
*Agust? Ubalde Bellot*
Chief Developer
C/ Comte Urgell 240 3A
Barcelona 08036
aubalde at presenceco.com

Ph: +34 93 10 10 300
Fx: +34 93 10 10 333

*www.presenceco.com* <http://www.presenceco.com/>

*Follow us on:*

*[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
<http://www.youtube.com/PresenceTechnology>*[image: in]*
<http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]*
<http://www.slideshare.net/PresenceTech>*[image: fb]*
<http://www.facebook.com/presencetech>

For additional information, please visit our website *www.presenceco.com*
<http://www.presenceco.com/>

-- 



*Presence Technology - DisclaimerThis message, its content and any file 
attached thereto is for the intended recipient only and is confidential and 
/or privileged. If you have received this e-mail in error or had access to 
it, you should note that the information in it is private and any use 
thereof is unauthorized. In such an event please notify us by e-mail or by 
telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by 
whatsoever means and any transmission or dissemination thereof to other 
persons is prohibited. It should be deleted immediately from your system. 
Presence Technology reserves the right to take legal action against any 
persons unlawfully gaining access to the content of any external message it 
has emitted.*

*For additional information, please visit our website **www.presenceco.com 
<http://www.presenceco.com/>*

 
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From krice at freeswitch.org  Mon Jun 20 15:38:34 2016
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 20 Jun 2016 06:38:34 -0500
Subject: [Freeswitch-users] Timestamp and timezone issues in CDR,
	bug or by design?
In-Reply-To: <CAE3mkNDWZ8fZuPCXFB=rQWYHFqyt7QES8NLOw6ah7P1_NhX+4Q@mail.gmail.com>
References: <CAE3mkNAwnmjRoYNtC22Na1QbzSf3xixy6e1eB0XBjpeZvNVFgA@mail.gmail.com>
	<80b8be5f-f941-626a-557e-9f2abbc03152@mst.edu>
	<CAE3mkNDWZ8fZuPCXFB=rQWYHFqyt7QES8NLOw6ah7P1_NhX+4Q@mail.gmail.com>
Message-ID: <9CDEB079-28F3-403B-BA8D-047BD271A773@freeswitch.org>

This generally depends on the provider. However if you are using a database such as postgresql, you can store time date stamps with a timezone and have your sql extract the tz you want to display. 

Sent from my iPhone

> On Jun 20, 2016, at 6:12 AM, vfclists <vfclists at gmail.com> wrote:
> 
> It is what I was thinking. I just wanted to be sure.
> 
> Do service providers generally present the CDRs using UTC time or do they adjust them to the customers time zone?
> 
>> On 20 June 2016 at 03:38, Nathan Neulinger <nneul at mst.edu> wrote:
>> epoch type timestamps are always UTC by definition, it's a universal/exact format.
>> 
>> -- Nathan
>> 
>> On 06/19/2016 08:41 PM, vfclists wrote:
>> >
>> > When I check the relationship between the epochs and the timestamps, eg answer_epoch vs answer_stamp I realize that
>> > after the daylight saving change the epochs are an hour behind. Is this by design? Are the timestamps supposed to be
>> > based on the computer's timezone by design, with the epochs based on the UTC value regardless of timezone and daylight
>> > savings?
>> >
>> > select
>> > (julianday('2016-03-30 15:23:00')-2440587.5) * 86400.0, -- 1459351379.999998, answer_epoch is 1459347780
>> >  (julianday('2016-03-30 14:23:00')-2440587.5) * 86400.0, -- 1459347780.000012, value above - 1hr
>> > (julianday('2016-01-27 11:58:53')-2440587.5) * 86400 --1453895933.000001 - prior to daylight savings
>> >
>> > The first 2 are from the sqlite CDR and the last from the CSV file.
>> >
>> >
>> >
>> > --
>> > Frank Church
>> http://devblog.brahmancreations.com
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From matt at supportedbusiness.com  Mon Jun 20 16:58:39 2016
From: matt at supportedbusiness.com (Matt Broad)
Date: Mon, 20 Jun 2016 13:58:39 +0100
Subject: [Freeswitch-users] freetdm using apt-get
Message-ID: <CAEeV61iGd71CP51WPWVvqy+_vHXXkE9whpw-nnjwwk=XniCFMQ@mail.gmail.com>

Hi,

I have installed freeswitch using the latest branch as per the instructions
on
https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie#Debian8Jessie-latestreleasebranch

My question is how do I add the freetdm module?  It isn't listed when doing
an apt-cache search freeswitch.

thanks
Matt
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From cmrienzo at gmail.com  Mon Jun 20 16:59:32 2016
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Mon, 20 Jun 2016 08:59:32 -0400
Subject: [Freeswitch-users] Tone detection issue
In-Reply-To: <CAD_iGr-5T60b0RbYb0n5A53KsQak7sDwtnSkt0Jqxf8WmnBr5Q@mail.gmail.com>
References: <CAD_iGr-5T60b0RbYb0n5A53KsQak7sDwtnSkt0Jqxf8WmnBr5Q@mail.gmail.com>
Message-ID: <CAD-qRkie-zwv1ZWnN3KJr8HPt=LSo7XWxvQz5byLsn6-UAj1Zg@mail.gmail.com>

Use spandsp_start_tone_detect instead.  It's better suited for detecting
tones + cadence.

On Sun, Jun 19, 2016 at 10:58 PM, Alexandru Covalschi <568691 at gmail.com>
wrote:

> Hi list,
>
> I have an issue using tone_detect. The thing is I must detect two very
> short tones - 1400 Hz and 2300 Hz 100ms coming one after another (with 100
> ms pause I suppose) after ~2 seconds the call is answered. I have pretty
> accurate wav file, in audacity I see pikes are 1400 and 2296.
> I've seen that in spandsp sources it is said that accuracy is ~3% so that
> should be enough,
> However tones aren't detected even if I set:
>
> <action application="tone_detect" data="tone_1_0 1395 r 0 set
> tone_first=true 1"/>
> <action application="tone_detect" data="tone_2_0 2295 r 0 set
> tone_second=true 1"/>
> <action application="tone_detect" data="tone_1_1 1396 r 0 set
> tone_first=true 1"/>
> <action application="tone_detect" data="tone_2_1 2296 r 0 set
> tone_second=true 1"/>
> <action application="tone_detect" data="tone_1_2 1397 r 0 set
> tone_first=true 1"/>
> <action application="tone_detect" data="tone_2_2 2297 r 0 set
> tone_second=true 1"/>
> <action application="tone_detect" data="tone_1_3 1398 r 0 set
> tone_first=true 1"/>
> <action application="tone_detect" data="tone_2_3 2298 r 0 set
> tone_second=true 1"/>
> <action application="tone_detect" data="tone_1_4 1399 r 0 set
> tone_first=true 1"/>
> <action application="tone_detect" data="tone_2_4 2299 r 0 set
> tone_second=true 1"/>
> <action application="tone_detect" data="tone_1_5 1400 r 0 set
> tone_first=true 1"/>
> <action application="tone_detect" data="tone_2_5 2300 r 0 set
> tone_second=true 1"/>
> <action application="tone_detect" data="tone_1_6 1401 r 0 set
> tone_first=true 1"/>
> <action application="tone_detect" data="tone_2_6 2301 r 0 set
> tone_second=true 1"/>
> <action application="tone_detect" data="tone_1_7 1402 r 0 set
> tone_first=true 1"/>
> <action application="tone_detect" data="tone_2_7 2302 r 0 set
> tone_second=true 1"/>
>
> Tried to do same using spandsp_start_tone_detect(), but I have no idea how
> to edit spandsp.conf to add a tone matching my needs.
>
> Can someone give me any clue? Any parameter I need to change in sources to
> obtain what I need?
>
> Debian Jessie
> FreeSWITCH version: 1.6.6-13-d2d0b32~64bit (-13-d2d0b32 64bit)
>
>
> Thanks in advance!
> --
> Alexandru Covalschi
> ABRISS-Solutions
> VoIP engineer and system administrator
> tel: +37367398493
> web: http://abriss.solutions/ <http://abs-telecom.com/>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fanx07 at gmail.com  Mon Jun 20 17:15:16 2016
From: fanx07 at gmail.com (Anonim Stefan)
Date: Mon, 20 Jun 2016 16:15:16 +0300
Subject: [Freeswitch-users] Delete previously appended header for the
	bridged leg
Message-ID: <CAO7wsJsBWBV43qvEs_TBJtX654ZC3dOizXR6osa1e8+oKOZYAg@mail.gmail.com>

Hi,

I issue this command via python ESL:
"
originate{origination_uuid=bla}{sip_h_X-HEADER1=VALUE1}sofia/gateway/gateway1/to_uri1
&bridge({sip_h_X-HEADER2=VALUE2}sofia/gateway/gateway1/to_uri2)
"

The problem I have is that HEADER1 is still existent in the bridged leg. I
have tried adding &unset(sip_h_X-HEADER1) without success.

Also I've tried with [] as said here [1], without succes.

What would be the solution to remove the sip_h_X-HEADER1 for the bridged
party?

[1]
https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-ChannelVariablesinDialstrings

Thank you,
Stefan
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From benjamin.cropley at gmail.com  Mon Jun 20 17:21:38 2016
From: benjamin.cropley at gmail.com (Benjamin Cropley)
Date: Mon, 20 Jun 2016 14:21:38 +0100
Subject: [Freeswitch-users] 2 calls to the same extension
In-Reply-To: <CAKThY07NVaJmcOd=CtghiihFnBW=jsxbkUPF+8tTCS2nawVJrg@mail.gmail.com>
References: <CAKThY07NVaJmcOd=CtghiihFnBW=jsxbkUPF+8tTCS2nawVJrg@mail.gmail.com>
Message-ID: <CAF7jizJ6RdNLeozxUrAjkNb8740RV_P56wtzVpw3HG09XMvynQ@mail.gmail.com>

Check out https://tools.ietf.org/html/rfc5359#section-2.11

(not sure if FreeSWITCH actually accepts Join headers, and probably relies
on presence information relating to the call being shared, but this is a
starting point)

On Mon, Jun 20, 2016 at 12:14 PM, Agust? Ubalde <aubalde at presenceco.com>
wrote:

> Hi all,
>
> Would be possible to bridge 2 calls about the same local extension? Or is
> it better to use a conference?
>
>
> Thanks!
>
> *PRESENCE TECHNOLOGY*
> *Agust? Ubalde Bellot*
> Chief Developer
> C/ Comte Urgell 240 3A
> Barcelona 08036
> aubalde at presenceco.com
>
> Ph: +34 93 10 10 300
> Fx: +34 93 10 10 333
>
> *www.presenceco.com* <http://www.presenceco.com/>
>
> *Follow us on:*
>
> *[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
> <http://www.youtube.com/PresenceTechnology>*[image: in]*
> <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]*
> <http://www.slideshare.net/PresenceTech>*[image: fb]*
> <http://www.facebook.com/presencetech>
>
> For additional information, please visit our website *www.presenceco.com*
> <http://www.presenceco.com/>
>
>
> *Presence Technology - DisclaimerThis message, its content and any file
> attached thereto is for the intended recipient only and is confidential and
> /or privileged. If you have received this e-mail in error or had access to
> it, you should note that the information in it is private and any use
> thereof is unauthorized. In such an event please notify us by e-mail or by
> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by
> whatsoever means and any transmission or dissemination thereof to other
> persons is prohibited. It should be deleted immediately from your system.
> Presence Technology reserves the right to take legal action against any
> persons unlawfully gaining access to the content of any external message it
> has emitted.*
>
> *For additional information, please visit our website **www.presenceco.com
> <http://www.presenceco.com/>*
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From ashwinrkjain at gmail.com  Mon Jun 20 17:57:53 2016
From: ashwinrkjain at gmail.com (Ashwin Jain)
Date: Mon, 20 Jun 2016 19:27:53 +0530
Subject: [Freeswitch-users] Audio skip issue with simultaneous ringing
In-Reply-To: <8F5AA30D-C8FD-48BA-91B0-141FCCA286BF@kavun.ch>
References: <CAM9SNVTQJo5jCZ+jb9Yy6fErb7UdUtH1QpbNn=8kPkdaP=m0vg@mail.gmail.com>
	<CAM9SNVRpSMnm+fdF9Zq9haZMf3SYtQW99q2HxM0eEzz6wHQbdA@mail.gmail.com>
	<8F5AA30D-C8FD-48BA-91B0-141FCCA286BF@kavun.ch>
Message-ID: <CAM9SNVQN+T2jeJq0_JVAShJweodMtgPfsNFaQ+rZvUJUfes8ZQ@mail.gmail.com>

Hi Emrah,

Thanks for replying. We are using packed version (on ubuntu 14.04) hosted
on AWS (with modified kernel timer to 1000Hz).
It is m3.medium (Intel Xeon E5-2670 v2 with 4GB RAM and SSD). Any help
would be really helpful.

Bridge string is: [leg_progress_timeout=20,leg_timeout=20,presence_id=user1@
<domain>,bridge_answer_timeout=20]sofia/mgprofile/user1@
<domain>:5060,[leg_progress_timeout=20,leg_timeout=20,presence_id=user2@
<domain>,bridge_answer_timeout=20]sofia/mgprofile/user2@
<domain>:5060,[leg_progress_timeout=20,leg_timeout=20,presence_id=user3@
<domain>,bridge_answer_timeout=20]sofia/mgprofile/user3@
<domain>:5060,[leg_progress_timeout=20,leg_timeout=20,presence_id=user4@
<domain>,bridge_answer_timeout=20]sofia/mgprofile/user4@
<domain>:5060,[leg_progress_timeout=20,leg_timeout=20,presence_id=user5@
<domain>,bridge_answer_timeout=20]sofia/mgprofile/user5@
<domain>:5060,[leg_progress_timeout=20,leg_timeout=20,presence_id=user6@
<domain>,bridge_answer_timeout=20]sofia/mgprofile/user6@
<domain>:5060,[leg_progress_timeout=20,leg_timeout=20,presence_id=user7@
<domain>,bridge_answer_timeout=20]sofia/mgprofile/user7@
<domain>:5060,[leg_progress_timeout=20,leg_timeout=20,presence_id=user8@
<domain>,bridge_answer_timeout=20]sofia/mgprofile/user8@<domain>:5060

Logs are

2016-06-17 09:53:48.294515 [DEBUG] switch_channel.c:3659 Send signal
sofia/<profile_name>/<caller_id>@<caller_ip> [BREAK]
2016-06-17 09:53:48.294515 [NOTICE] sofia.c:7525 Channel
[sofia/<profile_name>/user1@<domain>:5060] has been answered
2016-06-17 09:53:48.294515 [DEBUG] switch_channel.c:3713
(sofia/<profile_name>/user1@<domain>:5060) Callstate Change RINGING ->
ACTIVE
2016-06-17 09:53:48.294515 [DEBUG] switch_core_codec.c:246
sofia/<profile_name>/<caller_id>@<caller_ip> Restore previous codec PCMU:0.
2016-06-17 09:53:48.294515 [NOTICE] switch_ivr_originate.c:3519 Hangup
sofia/<profile_name>/user2@<domain>:5060 [CS_CONSUME_MEDIA] [LOSE_RACE]
2016-06-17 09:53:48.294515 [DEBUG] switch_channel.c:3244 Send signal
sofia/<profile_name>/user2@<domain>:5060 [KILL]
2016-06-17 09:53:48.294515 [DEBUG] switch_core_session.c:1397 Send signal
sofia/<profile_name>/user2@<domain>:5060 [BREAK]
2016-06-17 09:53:48.294515 [NOTICE] switch_ivr_originate.c:3519 Hangup
sofia/<profile_name>/user3@<domain>:5060 [CS_CONSUME_MEDIA] [LOSE_RACE]
2016-06-17 09:53:48.294515 [DEBUG] switch_channel.c:3244 Send signal
sofia/<profile_name>/user3@<domain>:5060 [KILL]
2016-06-17 09:53:48.294515 [DEBUG] switch_core_session.c:1397 Send signal
sofia/<profile_name>/user3@<domain>:5060 [BREAK]
2016-06-17 09:53:48.294515 [NOTICE] switch_ivr_originate.c:3519 Hangup
sofia/<profile_name>/user4@<domain>:5060 [CS_CONSUME_MEDIA] [LOSE_RACE]
2016-06-17 09:53:48.294515 [DEBUG] switch_channel.c:3244 Send signal
sofia/<profile_name>/user4@<domain>:5060 [KILL]
2016-06-17 09:53:48.294515 [DEBUG] switch_core_session.c:1397 Send signal
sofia/<profile_name>/user4@<domain>:5060 [BREAK]
2016-06-17 09:53:48.294515 [NOTICE] switch_ivr_originate.c:3519 Hangup
sofia/<profile_name>/user5@<domain>:5060 [CS_CONSUME_MEDIA] [LOSE_RACE]
2016-06-17 09:53:48.294515 [DEBUG] switch_channel.c:3244 Send signal
sofia/<profile_name>/user5@<domain>:5060 [KILL]
2016-06-17 09:53:48.294515 [DEBUG] switch_core_session.c:1397 Send signal
sofia/<profile_name>/user5@<domain>:5060 [BREAK]
2016-06-17 09:53:48.294515 [NOTICE] switch_ivr_originate.c:3519 Hangup
sofia/<profile_name>/user6@<domain>:5060 [CS_CONSUME_MEDIA] [LOSE_RACE]
2016-06-17 09:53:48.294515 [DEBUG] switch_channel.c:3244 Send signal
sofia/<profile_name>/user6@<domain>:5060 [KILL]
2016-06-17 09:53:48.294515 [DEBUG] switch_core_session.c:1397 Send signal
sofia/<profile_name>/user6@<domain>:5060 [BREAK]
2016-06-17 09:53:48.294515 [NOTICE] switch_ivr_originate.c:3519 Hangup
sofia/<profile_name>/user7@<domain>:5060 [CS_CONSUME_MEDIA] [LOSE_RACE]
2016-06-17 09:53:48.294515 [DEBUG] switch_channel.c:3244 Send signal
sofia/<profile_name>/user7@<domain>:5060 [KILL]
2016-06-17 09:53:48.294515 [DEBUG] switch_core_session.c:1397 Send signal
sofia/<profile_name>/user7@<domain>:5060 [BREAK]
2016-06-17 09:53:48.294515 [NOTICE] switch_ivr_originate.c:3519 Hangup
sofia/<profile_name>/user8@<domain>:5060 [CS_CONSUME_MEDIA] [LOSE_RACE]
2016-06-17 09:53:48.294515 [DEBUG] switch_channel.c:3244 Send signal
sofia/<profile_name>/user8@<domain>:5060 [KILL]
2016-06-17 09:53:48.294515 [DEBUG] switch_core_session.c:1397 Send signal
sofia/<profile_name>/user8@<domain>:5060 [BREAK]
2016-06-17 09:53:48.294515 [DEBUG] switch_core_state_machine.c:472
(sofia/<profile_name>/user2@<domain>:5060) Running State Change CS_HANGUP
2016-06-17 09:53:48.294515 [DEBUG] switch_core_state_machine.c:735
(sofia/<profile_name>/user2@<domain>:5060) Callstate Change RINGING ->
HANGUP
2016-06-17 09:53:48.294515 [DEBUG] switch_core_state_machine.c:737
(sofia/<profile_name>/user2@<domain>:5060) State HANGUP
2016-06-17 09:53:48.294515 [DEBUG] mod_sofia.c:413 Channel
sofia/<profile_name>/user2@<domain>:5060 hanging up, cause: LOSE_RACE
2016-06-17 09:53:48.354483 [DEBUG] sofia.c:1324 Channel is already hungup.
2016-06-17 09:53:48.354483 [DEBUG] switch_core_state_machine.c:472
(sofia/<profile_name>/user6@<domain>:5060) Running State Change CS_HANGUP
2016-06-17 09:53:48.354483 [DEBUG] switch_core_state_machine.c:735
(sofia/<profile_name>/user6@<domain>:5060) Callstate Change DOWN -> HANGUP
2016-06-17 09:53:48.354483 [DEBUG] switch_core_state_machine.c:737
(sofia/<profile_name>/user6@<domain>:5060) State HANGUP
2016-06-17 09:53:48.354483 [DEBUG] mod_sofia.c:413 Channel
sofia/<profile_name>/user6@<domain>:5060 hanging up, cause: LOSE_RACE
2016-06-17 09:53:48.734452 [DEBUG] sofia.c:1324 Channel is already hungup.
2016-06-17 09:53:48.734452 [DEBUG] switch_core_state_machine.c:472
(sofia/<profile_name>/user7@<domain>:5060) Running State Change CS_HANGUP
2016-06-17 09:53:48.734452 [DEBUG] switch_core_state_machine.c:735
(sofia/<profile_name>/user7@<domain>:5060) Callstate Change DOWN -> HANGUP
2016-06-17 09:53:48.734452 [DEBUG] switch_core_state_machine.c:737
(sofia/<profile_name>/user7@<domain>:5060) State HANGUP
2016-06-17 09:53:48.734452 [DEBUG] mod_sofia.c:413 Channel
sofia/<profile_name>/user7@<domain>:5060 hanging up, cause: LOSE_RACE
2016-06-17 09:53:48.820119 [DEBUG] mod_sofia.c:476 Sending CANCEL to
sofia/<profile_name>/user7@<domain>:5060
2016-06-17 09:53:48.820119 [DEBUG] switch_core_state_machine.c:60
sofia/<profile_name>/user7@<domain>:5060 Standard HANGUP, cause: LOSE_RACE
2016-06-17 09:53:48.820119 [DEBUG] switch_core_state_machine.c:737
(sofia/<profile_name>/user7@<domain>:5060) State HANGUP going to sleep
2016-06-17 09:53:48.820119 [DEBUG] switch_core_state_machine.c:504
(sofia/<profile_name>/user7@<domain>:5060) State Change CS_HANGUP ->
CS_REPORTING
2016-06-17 09:53:48.820119 [DEBUG] switch_core_session.c:1397 Send signal
sofia/<profile_name>/user7@<domain>:5060 [BREAK]
2016-06-17 09:53:48.820119 [DEBUG] switch_core_state_machine.c:472
(sofia/<profile_name>/user7@<domain>:5060) Running State Change CS_REPORTING
2016-06-17 09:53:48.820119 [DEBUG] switch_core_state_machine.c:823
(sofia/<profile_name>/user7@<domain>:5060) State REPORTING
2016-06-17 09:53:48.820119 [DEBUG] switch_core_state_machine.c:104
sofia/<profile_name>/user7@<domain>:5060 Standard REPORTING, cause:
LOSE_RACE
2016-06-17 09:53:48.820119 [DEBUG] switch_core_state_machine.c:823
(sofia/<profile_name>/user7@<domain>:5060) State REPORTING going to sleep
2016-06-17 09:53:48.820119 [DEBUG] switch_core_state_machine.c:498
(sofia/<profile_name>/user7@<domain>:5060) State Change CS_REPORTING ->
CS_DESTROY
2016-06-17 09:53:48.820119 [DEBUG] switch_core_session.c:1397 Send signal
sofia/<profile_name>/user7@<domain>:5060 [BREAK]
2016-06-17 09:53:48.820119 [DEBUG] switch_core_session.c:1624 Session 243
(sofia/<profile_name>/user7@<domain>:5060) Locked, Waiting on external
entities
2016-06-17 09:53:49.174437 [DEBUG] mod_sofia.c:476 Sending CANCEL to
sofia/<profile_name>/user2@<domain>:5060
2016-06-17 09:53:49.174437 [DEBUG] switch_core_state_machine.c:60
sofia/<profile_name>/user2@<domain>:5060 Standard HANGUP, cause: LOSE_RACE
2016-06-17 09:53:49.174437 [DEBUG] switch_core_state_machine.c:737
(sofia/<profile_name>/user2@<domain>:5060) State HANGUP going to sleep
2016-06-17 09:53:49.174437 [DEBUG] switch_core_state_machine.c:504
(sofia/<profile_name>/user2@<domain>:5060) State Change CS_HANGUP ->
CS_REPORTING
2016-06-17 09:53:49.174437 [DEBUG] switch_core_session.c:1397 Send signal
sofia/<profile_name>/user2@<domain>:5060 [BREAK]
2016-06-17 09:53:49.174437 [DEBUG] switch_core_state_machine.c:472
(sofia/<profile_name>/user2@<domain>:5060) Running State Change CS_REPORTING
2016-06-17 09:53:49.174437 [DEBUG] switch_core_state_machine.c:823
(sofia/<profile_name>/user2@<domain>:5060) State REPORTING
2016-06-17 09:53:49.174437 [DEBUG] switch_core_state_machine.c:104
sofia/<profile_name>/user2@<domain>:5060 Standard REPORTING, cause:
LOSE_RACE
2016-06-17 09:53:49.174437 [DEBUG] switch_core_state_machine.c:823
(sofia/<profile_name>/user2@<domain>:5060) State REPORTING going to sleep
2016-06-17 09:53:49.174437 [DEBUG] switch_core_state_machine.c:498
(sofia/<profile_name>/user2@<domain>:5060) State Change CS_REPORTING ->
CS_DESTROY
2016-06-17 09:53:49.174437 [DEBUG] switch_core_session.c:1397 Send signal
sofia/<profile_name>/user2@<domain>:5060 [BREAK]
2016-06-17 09:53:49.174437 [DEBUG] switch_core_session.c:1624 Session 238
(sofia/<profile_name>/user2@<domain>:5060) Locked, Waiting on external
entities
2016-06-17 09:53:49.534437 [DEBUG] sofia.c:1324 Channel is already hungup.
2016-06-17 09:53:49.534437 [DEBUG] switch_core_state_machine.c:472
(sofia/<profile_name>/user5@<domain>:5060) Running State Change CS_HANGUP
2016-06-17 09:53:49.534437 [DEBUG] switch_core_state_machine.c:735
(sofia/<profile_name>/user5@<domain>:5060) Callstate Change DOWN -> HANGUP
2016-06-17 09:53:49.534437 [DEBUG] switch_core_state_machine.c:737
(sofia/<profile_name>/user5@<domain>:5060) State HANGUP
2016-06-17 09:53:49.534437 [DEBUG] mod_sofia.c:413 Channel
sofia/<profile_name>/user5@<domain>:5060 hanging up, cause: LOSE_RACE
2016-06-17 09:53:49.614437 [DEBUG] mod_sofia.c:476 Sending CANCEL to
sofia/<profile_name>/user5@<domain>:5060
2016-06-17 09:53:49.614437 [DEBUG] switch_core_state_machine.c:60
sofia/<profile_name>/user5@<domain>:5060 Standard HANGUP, cause: LOSE_RACE
2016-06-17 09:53:49.614437 [DEBUG] switch_core_state_machine.c:737
(sofia/<profile_name>/user5@<domain>:5060) State HANGUP going to sleep
2016-06-17 09:53:49.614437 [DEBUG] switch_core_state_machine.c:504
(sofia/<profile_name>/user5@<domain>:5060) State Change CS_HANGUP ->
CS_REPORTING
2016-06-17 09:53:49.614437 [DEBUG] switch_core_session.c:1397 Send signal
sofia/<profile_name>/user5@<domain>:5060 [BREAK]
2016-06-17 09:53:49.614437 [DEBUG] switch_core_state_machine.c:472
(sofia/<profile_name>/user5@<domain>:5060) Running State Change CS_REPORTING
2016-06-17 09:53:49.614437 [DEBUG] switch_core_state_machine.c:823
(sofia/<profile_name>/user5@<domain>:5060) State REPORTING
2016-06-17 09:53:49.614437 [DEBUG] switch_core_state_machine.c:104
sofia/<profile_name>/user5@<domain>:5060 Standard REPORTING, cause:
LOSE_RACE
2016-06-17 09:53:49.614437 [DEBUG] switch_core_state_machine.c:823
(sofia/<profile_name>/user5@<domain>:5060) State REPORTING going to sleep
2016-06-17 09:53:49.614437 [DEBUG] switch_core_state_machine.c:498
(sofia/<profile_name>/user5@<domain>:5060) State Change CS_REPORTING ->
CS_DESTROY
2016-06-17 09:53:49.614437 [DEBUG] switch_core_session.c:1397 Send signal
sofia/<profile_name>/user5@<domain>:5060 [BREAK]
2016-06-17 09:53:49.614437 [DEBUG] switch_core_session.c:1624 Session 241
(sofia/<profile_name>/user5@<domain>:5060) Locked, Waiting on external
entities
2016-06-17 09:53:49.974435 [DEBUG] sofia.c:1324 Channel is already hungup.
2016-06-17 09:53:49.974435 [DEBUG] switch_core_state_machine.c:472
(sofia/<profile_name>/user4@<domain>:5060) Running State Change CS_HANGUP
2016-06-17 09:53:49.974435 [DEBUG] switch_core_state_machine.c:735
(sofia/<profile_name>/user4@<domain>:5060) Callstate Change DOWN -> HANGUP
2016-06-17 09:53:49.974435 [DEBUG] switch_core_state_machine.c:737
(sofia/<profile_name>/user4@<domain>:5060) State HANGUP
2016-06-17 09:53:49.974435 [DEBUG] mod_sofia.c:413 Channel
sofia/<profile_name>/user4@<domain>:5060 hanging up, cause: LOSE_RACE
2016-06-17 09:53:50.054437 [DEBUG] mod_sofia.c:476 Sending CANCEL to
sofia/<profile_name>/user4@<domain>:5060
2016-06-17 09:53:50.054437 [DEBUG] switch_core_state_machine.c:60
sofia/<profile_name>/user4@<domain>:5060 Standard HANGUP, cause: LOSE_RACE
2016-06-17 09:53:50.054437 [DEBUG] switch_core_state_machine.c:737
(sofia/<profile_name>/user4@<domain>:5060) State HANGUP going to sleep
2016-06-17 09:53:50.054437 [DEBUG] switch_core_state_machine.c:504
(sofia/<profile_name>/user4@<domain>:5060) State Change CS_HANGUP ->
CS_REPORTING
2016-06-17 09:53:50.054437 [DEBUG] switch_core_session.c:1397 Send signal
sofia/<profile_name>/user4@<domain>:5060 [BREAK]
2016-06-17 09:53:50.054437 [DEBUG] switch_core_state_machine.c:472
(sofia/<profile_name>/user4@<domain>:5060) Running State Change CS_REPORTING
2016-06-17 09:53:50.054437 [DEBUG] switch_core_state_machine.c:823
(sofia/<profile_name>/user4@<domain>:5060) State REPORTING
2016-06-17 09:53:50.054437 [DEBUG] switch_core_state_machine.c:104
sofia/<profile_name>/user4@<domain>:5060 Standard REPORTING, cause:
LOSE_RACE
2016-06-17 09:53:50.054437 [DEBUG] switch_core_state_machine.c:823
(sofia/<profile_name>/user4@<domain>:5060) State REPORTING going to sleep
2016-06-17 09:53:50.054437 [DEBUG] switch_core_state_machine.c:498
(sofia/<profile_name>/user4@<domain>:5060) State Change CS_REPORTING ->
CS_DESTROY
2016-06-17 09:53:50.054437 [DEBUG] switch_core_session.c:1397 Send signal
sofia/<profile_name>/user4@<domain>:5060 [BREAK]
2016-06-17 09:53:50.054437 [DEBUG] switch_core_session.c:1624 Session 240
(sofia/<profile_name>/user4@<domain>:5060) Locked, Waiting on external
entities
2016-06-17 09:53:50.394437 [DEBUG] mod_sofia.c:476 Sending CANCEL to
sofia/<profile_name>/user6@<domain>:5060
2016-06-17 09:53:50.394437 [DEBUG] switch_core_state_machine.c:60
sofia/<profile_name>/user6@<domain>:5060 Standard HANGUP, cause: LOSE_RACE
2016-06-17 09:53:50.394437 [DEBUG] switch_core_state_machine.c:737
(sofia/<profile_name>/user6@<domain>:5060) State HANGUP going to sleep
2016-06-17 09:53:50.394437 [DEBUG] switch_core_state_machine.c:504
(sofia/<profile_name>/user6@<domain>:5060) State Change CS_HANGUP ->
CS_REPORTING
2016-06-17 09:53:50.394437 [DEBUG] switch_core_session.c:1397 Send signal
sofia/<profile_name>/user6@<domain>:5060 [BREAK]
2016-06-17 09:53:50.394437 [DEBUG] switch_core_state_machine.c:472
(sofia/<profile_name>/user6@<domain>:5060) Running State Change CS_REPORTING
2016-06-17 09:53:50.394437 [DEBUG] switch_core_state_machine.c:823
(sofia/<profile_name>/user6@<domain>:5060) State REPORTING
2016-06-17 09:53:50.394437 [DEBUG] switch_core_state_machine.c:104
sofia/<profile_name>/user6@<domain>:5060 Standard REPORTING, cause:
LOSE_RACE
2016-06-17 09:53:50.394437 [DEBUG] switch_core_state_machine.c:823
(sofia/<profile_name>/user6@<domain>:5060) State REPORTING going to sleep
2016-06-17 09:53:50.394437 [DEBUG] switch_core_state_machine.c:498
(sofia/<profile_name>/user6@<domain>:5060) State Change CS_REPORTING ->
CS_DESTROY
2016-06-17 09:53:50.394437 [DEBUG] switch_core_session.c:1397 Send signal
sofia/<profile_name>/user6@<domain>:5060 [BREAK]
2016-06-17 09:53:50.394437 [DEBUG] switch_core_session.c:1624 Session 242
(sofia/<profile_name>/user6@<domain>:5060) Locked, Waiting on external
entities
2016-06-17 09:53:50.754437 [DEBUG] mod_sofia.c:780 Local SDP
sofia/<profile_name>/<caller_id>@<caller_ip>:
v=0
o=FreeSWITCH 1466150805 1466150807 IN IP4 <fs_public_ip>
s=FreeSWITCH
c=IN IP4 <fs_public_ip>
t=0 0
m=audio 31614 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

2016-06-17 09:53:50.754437 [DEBUG] switch_core_session.c:1062 Send signal
sofia/<profile_name>/<caller_id>@<caller_ip> [BREAK]
2016-06-17 09:53:50.754437 [DEBUG] sofia.c:6658 Channel
sofia/<profile_name>/<caller_id>@<caller_ip> entering state [completed][200]
2016-06-17 09:53:50.754437 [DEBUG] switch_core_session.c:913 Send signal
sofia/<profile_name>/<caller_id>@<caller_ip> [BREAK]
2016-06-17 09:53:50.754437 [NOTICE] switch_ivr_originate.c:3546 Channel
[sofia/<profile_name>/<caller_id>@<caller_ip>] has been answered
2016-06-17 09:53:50.754437 [DEBUG] switch_channel.c:3713
(sofia/<profile_name>/<caller_id>@<caller_ip>) Callstate Change EARLY ->
ACTIVE
2016-06-17 09:53:50.754437 [DEBUG] switch_ivr_originate.c:3604 Originate
Resulted in Success: [sofia/<profile_name>/user1@<domain>:5060]
2016-06-17 09:53:50.754437 [DEBUG] switch_channel.c:1230
sofia/<profile_name>/<caller_id>@<caller_ip> EXPORTING[bridge_export_vars]
[hold_music]=[sounds/moh.wav] to sofia/<profile_name>/user1@<domain>:5060
2016-06-17 09:53:50.754437 [DEBUG] switch_core_session.c:913 Send signal
sofia/<profile_name>/user1@<domain>:5060 [BREAK]
2016-06-17 09:53:50.754437 [DEBUG] switch_core_session.c:913 Send signal
sofia/<profile_name>/<caller_id>@<caller_ip> [BREAK]
2016-06-17 09:53:50.754437 [DEBUG] switch_ivr_bridge.c:1473
(sofia/<profile_name>/user1@<domain>:5060) State Change CS_CONSUME_MEDIA ->
CS_EXCHANGE_MEDIA
2016-06-17 09:53:50.754437 [DEBUG] switch_core_session.c:1397 Send signal
sofia/<profile_name>/user1@<domain>:5060 [BREAK]
2016-06-17 09:53:50.754437 [NOTICE] switch_core_session.c:1642 Session 238
(sofia/<profile_name>/user2@<domain>:5060) Ended
2016-06-17 09:53:50.754437 [NOTICE] switch_core_session.c:1646 Close
Channel sofia/<profile_name>/user2@<domain>:5060 [CS_DESTROY]
2016-06-17 09:53:50.754437 [DEBUG] switch_core_state_machine.c:626
(sofia/<profile_name>/user2@<domain>:5060) Running State Change CS_DESTROY

On Mon, Jun 20, 2016 at 3:59 PM, Emrah <lists at kavun.ch> wrote:

> Hello,
> Can you share your bridge string as well as some logs?
> What kind of processor are you running this on?
> Did you source compile your FS or are you using a packaged version?
>
> Emrah
>
> On Jun 19, 2016, at 5:36 PM, Ashwin Jain <ashwinrkjain at gmail.com> wrote:
>
> Any input on this would be really helpful.
>
> On Sat, Jun 18, 2016 at 12:52 PM, Ashwin Jain <ashwinrkjain at gmail.com>
> wrote:
>
>> Hi all,
>>
>> We are running Freeswitch version 1.4.26. We are facing a major problem
>> related to audio skip with simultaneous ringing. In one flow we are ringing
>> around 5 agents at the same time. Whenever agent 1 (any one of the agent)
>> picks up the call, other legs (to other agents) gets a hangup event. Due to
>> this, hangup event to all the other agent "CALL_ANSWER" event on the other
>> side gets delayed by around 2-3 seconds and generally caller missed the
>> first word from the agent.
>>
>> Event 1: Call Answered by agent 1
>> Event 2: Hangup for agent 2 (due to LOSE_RACE)
>> Event 3: Hangup for agent 3 (due to LOSE_RACE)
>> Event 4: Hangup for agent 4 (due to LOSE_RACE)
>> Event 5: Hangup for agent 5 (due to LOSE_RACE)
>> Event 6: Call answered to caller
>>
>> Time different between Event 1 and Event 6 is about 2-3 seconds.
>> Any help would be really really helpful
>>
>> --
>> Thanks and Regards,
>> Ashwin Jain
>>
>
>
>
> --
>
> *Ashwin Jain*
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Ashwin Jain* / Director of Engineering

*Email** :* ashwin at metroguild.com <chiqeetaj at metroguild.com>

*Phone :* (415) 800-0834 / +91-97424-94330
*Website* : www.metroguild.com
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From 568691 at gmail.com  Mon Jun 20 18:58:45 2016
From: 568691 at gmail.com (Alexandru Covalschi)
Date: Mon, 20 Jun 2016 17:58:45 +0300
Subject: [Freeswitch-users] Subject: Re:  Tone detection issue
Message-ID: <CAD_iGr-+M5uzgbqj7cSkofkbZoVMg6-0=AyMbCBZSwn088-MQA@mail.gmail.com>

Thanks for your reply Christopher!

Can you please specify how should I edit spandsp.conf (tones section) to
add my tone? I was trying for hours but didn't get any result, I just can't
get the logic and I'm struggling.

Cheers,
Alex



> From: Christopher Rienzo <cmrienzo at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Mon, 20 Jun 2016 08:59:32 -0400
> Subject: Re: [Freeswitch-users] Tone detection issue
> Use spandsp_start_tone_detect instead.  It's better suited for detecting
> tones + cadence.
>
> On Sun, Jun 19, 2016 at 10:58 PM, Alexandru Covalschi <568691 at gmail.com>
> wrote:
>
>> Hi list,
>>
>> I have an issue using tone_detect. The thing is I must detect two very
>> short tones - 1400 Hz and 2300 Hz 100ms coming one after another (with 100
>> ms pause I suppose) after ~2 seconds the call is answered. I have pretty
>> accurate wav file, in audacity I see pikes are 1400 and 2296.
>> I've seen that in spandsp sources it is said that accuracy is ~3% so that
>> should be enough,
>> However tones aren't detected even if I set:
>>
>> <action application="tone_detect" data="tone_1_0 1395 r 0 set
>> tone_first=true 1"/>
>> <action application="tone_detect" data="tone_2_0 2295 r 0 set
>> tone_second=true 1"/>
>> <action application="tone_detect" data="tone_1_1 1396 r 0 set
>> tone_first=true 1"/>
>> <action application="tone_detect" data="tone_2_1 2296 r 0 set
>> tone_second=true 1"/>
>> <action application="tone_detect" data="tone_1_2 1397 r 0 set
>> tone_first=true 1"/>
>> <action application="tone_detect" data="tone_2_2 2297 r 0 set
>> tone_second=true 1"/>
>> <action application="tone_detect" data="tone_1_3 1398 r 0 set
>> tone_first=true 1"/>
>> <action application="tone_detect" data="tone_2_3 2298 r 0 set
>> tone_second=true 1"/>
>> <action application="tone_detect" data="tone_1_4 1399 r 0 set
>> tone_first=true 1"/>
>> <action application="tone_detect" data="tone_2_4 2299 r 0 set
>> tone_second=true 1"/>
>> <action application="tone_detect" data="tone_1_5 1400 r 0 set
>> tone_first=true 1"/>
>> <action application="tone_detect" data="tone_2_5 2300 r 0 set
>> tone_second=true 1"/>
>> <action application="tone_detect" data="tone_1_6 1401 r 0 set
>> tone_first=true 1"/>
>> <action application="tone_detect" data="tone_2_6 2301 r 0 set
>> tone_second=true 1"/>
>> <action application="tone_detect" data="tone_1_7 1402 r 0 set
>> tone_first=true 1"/>
>> <action application="tone_detect" data="tone_2_7 2302 r 0 set
>> tone_second=true 1"/>
>>
>> Tried to do same using spandsp_start_tone_detect(), but I have no idea
>> how to edit spandsp.conf to add a tone matching my needs.
>>
>> Can someone give me any clue? Any parameter I need to change in sources
>> to obtain what I need?
>>
>> Debian Jessie
>> FreeSWITCH version: 1.6.6-13-d2d0b32~64bit (-13-d2d0b32 64bit)
>>
>>
>> Thanks in advance!
>> --
>> Alexandru Covalschi
>> ABRISS-Solutions
>> VoIP engineer and system administrator
>> tel: +37367398493
>> web: http://abriss.solutions/ <http://abs-telecom.com/>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> -
>


-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
tel: +37367398493
web: http://abriss.solutions/ <http://abs-telecom.com/>
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From mike at jerris.com  Mon Jun 20 19:34:29 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 20 Jun 2016 11:34:29 -0400
Subject: [Freeswitch-users] Audio skip issue with simultaneous ringing
In-Reply-To: <CAM9SNVTQJo5jCZ+jb9Yy6fErb7UdUtH1QpbNn=8kPkdaP=m0vg@mail.gmail.com>
References: <CAM9SNVTQJo5jCZ+jb9Yy6fErb7UdUtH1QpbNn=8kPkdaP=m0vg@mail.gmail.com>
Message-ID: <CAEwTsAjozSr2eQyPht1nxP5AophRUeqgxYaWtqBrn5+HA9o1zw@mail.gmail.com>

You are using older versions, I would first check if we have fixed this by
testing the latest releases.  1.4 is only receiving critical security
releases now and would not receive a fix if we found a problem.

On Saturday, June 18, 2016, Ashwin Jain <ashwinrkjain at gmail.com> wrote:

> Hi all,
>
> We are running Freeswitch version 1.4.26. We are facing a major problem
> related to audio skip with simultaneous ringing. In one flow we are ringing
> around 5 agents at the same time. Whenever agent 1 (any one of the agent)
> picks up the call, other legs (to other agents) gets a hangup event. Due to
> this, hangup event to all the other agent "CALL_ANSWER" event on the other
> side gets delayed by around 2-3 seconds and generally caller missed the
> first word from the agent.
>
> Event 1: Call Answered by agent 1
> Event 2: Hangup for agent 2 (due to LOSE_RACE)
> Event 3: Hangup for agent 3 (due to LOSE_RACE)
> Event 4: Hangup for agent 4 (due to LOSE_RACE)
> Event 5: Hangup for agent 5 (due to LOSE_RACE)
> Event 6: Call answered to caller
>
> Time different between Event 1 and Event 6 is about 2-3 seconds.
> Any help would be really really helpful
>
> --
> Thanks and Regards,
> Ashwin Jain
>
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From ashwinrkjain at gmail.com  Tue Jun 21 00:16:17 2016
From: ashwinrkjain at gmail.com (Ashwin Jain)
Date: Tue, 21 Jun 2016 01:46:17 +0530
Subject: [Freeswitch-users] Audio skip issue with simultaneous ringing
In-Reply-To: <CAEwTsAjozSr2eQyPht1nxP5AophRUeqgxYaWtqBrn5+HA9o1zw@mail.gmail.com>
References: <CAM9SNVTQJo5jCZ+jb9Yy6fErb7UdUtH1QpbNn=8kPkdaP=m0vg@mail.gmail.com>
	<CAEwTsAjozSr2eQyPht1nxP5AophRUeqgxYaWtqBrn5+HA9o1zw@mail.gmail.com>
Message-ID: <CAM9SNVSQSNp7gTfCV2gJ6t8BYmVeKFCP01dC9WN+b0-RENMMAg@mail.gmail.com>

Sure. We can update to the latest version. Can you confirm if this is the
issue, or is it some setup/configuration issue with our system?

On Mon, Jun 20, 2016 at 9:04 PM, Michael Jerris <mike at jerris.com> wrote:

> You are using older versions, I would first check if we have fixed this by
> testing the latest releases.  1.4 is only receiving critical security
> releases now and would not receive a fix if we found a problem.
>
> On Saturday, June 18, 2016, Ashwin Jain <ashwinrkjain at gmail.com> wrote:
>
>> Hi all,
>>
>> We are running Freeswitch version 1.4.26. We are facing a major problem
>> related to audio skip with simultaneous ringing. In one flow we are ringing
>> around 5 agents at the same time. Whenever agent 1 (any one of the agent)
>> picks up the call, other legs (to other agents) gets a hangup event. Due to
>> this, hangup event to all the other agent "CALL_ANSWER" event on the other
>> side gets delayed by around 2-3 seconds and generally caller missed the
>> first word from the agent.
>>
>> Event 1: Call Answered by agent 1
>> Event 2: Hangup for agent 2 (due to LOSE_RACE)
>> Event 3: Hangup for agent 3 (due to LOSE_RACE)
>> Event 4: Hangup for agent 4 (due to LOSE_RACE)
>> Event 5: Hangup for agent 5 (due to LOSE_RACE)
>> Event 6: Call answered to caller
>>
>> Time different between Event 1 and Event 6 is about 2-3 seconds.
>> Any help would be really really helpful
>>
>> --
>> Thanks and Regards,
>> Ashwin Jain
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Ashwin Jain* / Director of Engineering

*Email** :* ashwin at metroguild.com <chiqeetaj at metroguild.com>

*Phone :* (415) 800-0834 / +91-97424-94330
*Website* : www.metroguild.com
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From kathleen at freeswitch.org  Tue Jun 21 01:25:49 2016
From: kathleen at freeswitch.org (Kathleen King)
Date: Mon, 20 Jun 2016 14:25:49 -0700
Subject: [Freeswitch-users] Official ClueCon 2016 after party hosted by
	Telnyx!
Message-ID: <CAOStUPZfV0sxeYtvnEO-hQX_8=SmjwpPaQUCYedHhkZj4U7Ukg@mail.gmail.com>

[image: Banner] <#>

Come and enjoy drinks and apps on us, while networking with your peers and
enjoying live music, football, ping pong, and old school video games along
our 30? media wall.



Learn More <https://info.telnyx.com/cluecon>



[image: When Icon]

When

Wednesday, August 10, 2016 from 7:00 PM ? 12:00 AM
[image: Where Icon]

Where

311 W Superior St, Ste 504
Chicago, IL 60654

Located less than 2 miles from Swissotel, temporary shuttle service will be
available.
[image: What Icon]

What
 Drinks, Food, Games, Networking and Special Giveaways!
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From s.safarov at gmail.com  Tue Jun 21 08:14:42 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 21 Jun 2016 04:14:42 +0000
Subject: [Freeswitch-users] Delete previously appended header for the
 bridged leg
In-Reply-To: <CAO7wsJsBWBV43qvEs_TBJtX654ZC3dOizXR6osa1e8+oKOZYAg@mail.gmail.com>
References: <CAO7wsJsBWBV43qvEs_TBJtX654ZC3dOizXR6osa1e8+oKOZYAg@mail.gmail.com>
Message-ID: <CAHtxdDeXjx30gvt+C63vQad+L5jNLZ4OP2nCtcSwfyCD=we=bw@mail.gmail.com>

Try
originate{origination_uuid=bla}{sip_h_X-HEADER1=VALUE1}sofia/gateway/gateway1/to_uri1
&bridge({sip_h_X-HEADER1=_undef_,sip_h_X-HEADER2=
VALUE2}sofia/gateway/gateway1/to_uri2)

??, 20 ???. 2016 ?. ? 16:16, Anonim Stefan <fanx07 at gmail.com>:

> Hi,
>
> I issue this command via python ESL:
> "
> originate{origination_uuid=bla}{sip_h_X-HEADER1=VALUE1}sofia/gateway/gateway1/to_uri1
> &bridge({sip_h_X-HEADER2=VALUE2}sofia/gateway/gateway1/to_uri2)
> "
>
> The problem I have is that HEADER1 is still existent in the bridged leg. I
> have tried adding &unset(sip_h_X-HEADER1) without success.
>
> Also I've tried with [] as said here [1], without succes.
>
> What would be the solution to remove the sip_h_X-HEADER1 for the bridged
> party?
>
> [1]
> https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-ChannelVariablesinDialstrings
>
> Thank you,
> Stefan
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From nandy1925 at gmail.com  Tue Jun 21 16:46:48 2016
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Tue, 21 Jun 2016 20:46:48 +0800
Subject: [Freeswitch-users] Subject: Re: Tone detection issue
In-Reply-To: <CAD_iGr-+M5uzgbqj7cSkofkbZoVMg6-0=AyMbCBZSwn088-MQA@mail.gmail.com>
References: <CAD_iGr-+M5uzgbqj7cSkofkbZoVMg6-0=AyMbCBZSwn088-MQA@mail.gmail.com>
Message-ID: <CAHKqMb5WYu==9Y0F04PiXF6HKpUdztfQaBS437XGZUMZ+HGxZQ@mail.gmail.com>

Hi Alex,

Referring to this wiki:

https://wiki.freeswitch.org/wiki/Mod_spandsp#Call_Progress

You need to tune the detector (record the tone and make a freq analysis).
Take note the series of <element /> in spandsp.conf.  That's the sequence
of tones to be detected. The min/max is the duration of every tone.

In the dialplan app start_tone_detect, the data=1 refers to the <descriptor
name="1">. You can create your own set of progress tones.

Good luck.

/Nandy

On Mon, Jun 20, 2016 at 10:58 PM, Alexandru Covalschi <568691 at gmail.com>
wrote:

> Thanks for your reply Christopher!
>
> Can you please specify how should I edit spandsp.conf (tones section) to
> add my tone? I was trying for hours but didn't get any result, I just can't
> get the logic and I'm struggling.
>
> Cheers,
> Alex
>
>
>
>> From: Christopher Rienzo <cmrienzo at gmail.com>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Cc:
>> Date: Mon, 20 Jun 2016 08:59:32 -0400
>> Subject: Re: [Freeswitch-users] Tone detection issue
>> Use spandsp_start_tone_detect instead.  It's better suited for detecting
>> tones + cadence.
>>
>> On Sun, Jun 19, 2016 at 10:58 PM, Alexandru Covalschi <568691 at gmail.com>
>> wrote:
>>
>>> Hi list,
>>>
>>> I have an issue using tone_detect. The thing is I must detect two very
>>> short tones - 1400 Hz and 2300 Hz 100ms coming one after another (with 100
>>> ms pause I suppose) after ~2 seconds the call is answered. I have pretty
>>> accurate wav file, in audacity I see pikes are 1400 and 2296.
>>> I've seen that in spandsp sources it is said that accuracy is ~3% so
>>> that should be enough,
>>> However tones aren't detected even if I set:
>>>
>>> <action application="tone_detect" data="tone_1_0 1395 r 0 set
>>> tone_first=true 1"/>
>>> <action application="tone_detect" data="tone_2_0 2295 r 0 set
>>> tone_second=true 1"/>
>>> <action application="tone_detect" data="tone_1_1 1396 r 0 set
>>> tone_first=true 1"/>
>>> <action application="tone_detect" data="tone_2_1 2296 r 0 set
>>> tone_second=true 1"/>
>>> <action application="tone_detect" data="tone_1_2 1397 r 0 set
>>> tone_first=true 1"/>
>>> <action application="tone_detect" data="tone_2_2 2297 r 0 set
>>> tone_second=true 1"/>
>>> <action application="tone_detect" data="tone_1_3 1398 r 0 set
>>> tone_first=true 1"/>
>>> <action application="tone_detect" data="tone_2_3 2298 r 0 set
>>> tone_second=true 1"/>
>>> <action application="tone_detect" data="tone_1_4 1399 r 0 set
>>> tone_first=true 1"/>
>>> <action application="tone_detect" data="tone_2_4 2299 r 0 set
>>> tone_second=true 1"/>
>>> <action application="tone_detect" data="tone_1_5 1400 r 0 set
>>> tone_first=true 1"/>
>>> <action application="tone_detect" data="tone_2_5 2300 r 0 set
>>> tone_second=true 1"/>
>>> <action application="tone_detect" data="tone_1_6 1401 r 0 set
>>> tone_first=true 1"/>
>>> <action application="tone_detect" data="tone_2_6 2301 r 0 set
>>> tone_second=true 1"/>
>>> <action application="tone_detect" data="tone_1_7 1402 r 0 set
>>> tone_first=true 1"/>
>>> <action application="tone_detect" data="tone_2_7 2302 r 0 set
>>> tone_second=true 1"/>
>>>
>>> Tried to do same using spandsp_start_tone_detect(), but I have no idea
>>> how to edit spandsp.conf to add a tone matching my needs.
>>>
>>> Can someone give me any clue? Any parameter I need to change in sources
>>> to obtain what I need?
>>>
>>> Debian Jessie
>>> FreeSWITCH version: 1.6.6-13-d2d0b32~64bit (-13-d2d0b32 64bit)
>>>
>>>
>>> Thanks in advance!
>>> --
>>> Alexandru Covalschi
>>> ABRISS-Solutions
>>> VoIP engineer and system administrator
>>> tel: +37367398493
>>> web: http://abriss.solutions/ <http://abs-telecom.com/>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> -
>>
>
>
> --
> Alexandru Covalschi
> ABRISS-Solutions
> VoIP engineer and system administrator
> tel: +37367398493
> web: http://abriss.solutions/ <http://abs-telecom.com/>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From cmrienzo at gmail.com  Tue Jun 21 17:14:38 2016
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Tue, 21 Jun 2016 09:14:38 -0400
Subject: [Freeswitch-users] Subject: Re: Tone detection issue
In-Reply-To: <CAHKqMb5WYu==9Y0F04PiXF6HKpUdztfQaBS437XGZUMZ+HGxZQ@mail.gmail.com>
References: <CAD_iGr-+M5uzgbqj7cSkofkbZoVMg6-0=AyMbCBZSwn088-MQA@mail.gmail.com>
	<CAHKqMb5WYu==9Y0F04PiXF6HKpUdztfQaBS437XGZUMZ+HGxZQ@mail.gmail.com>
Message-ID: <CAD-qRkgtFT6AKVRjaWJW7ksgrC9taGam1javYGrX+_6+APngUg@mail.gmail.com>

Try something like this to start:

 <descriptors>
     <descriptor name="custom_foo">
       <tone name="CUSTOM_FOO">
         <element freq1="1400" freq2="0" min="80" max="120"/>
         <element freq1="0" freq2="0" min="80" max="120"/>
         <element freq1="2300" freq2="0" min="80" max="120"/>
       </tone>
     </descriptor>
</descriptors>

Enable debug tracing if it doesn't work and see how close you get.


On Tue, Jun 21, 2016 at 8:46 AM, Nandy Dagondon <nandy1925 at gmail.com> wrote:

> Hi Alex,
>
> Referring to this wiki:
>
> https://wiki.freeswitch.org/wiki/Mod_spandsp#Call_Progress
>
> You need to tune the detector (record the tone and make a freq analysis).
> Take note the series of <element /> in spandsp.conf.  That's the sequence
> of tones to be detected. The min/max is the duration of every tone.
>
> In the dialplan app start_tone_detect, the data=1 refers to the
> <descriptor name="1">. You can create your own set of progress tones.
>
> Good luck.
>
> /Nandy
>
> On Mon, Jun 20, 2016 at 10:58 PM, Alexandru Covalschi <568691 at gmail.com>
> wrote:
>
>> Thanks for your reply Christopher!
>>
>> Can you please specify how should I edit spandsp.conf (tones section) to
>> add my tone? I was trying for hours but didn't get any result, I just can't
>> get the logic and I'm struggling.
>>
>> Cheers,
>> Alex
>>
>>
>>
>>> From: Christopher Rienzo <cmrienzo at gmail.com>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Cc:
>>> Date: Mon, 20 Jun 2016 08:59:32 -0400
>>> Subject: Re: [Freeswitch-users] Tone detection issue
>>> Use spandsp_start_tone_detect instead.  It's better suited for detecting
>>> tones + cadence.
>>>
>>> On Sun, Jun 19, 2016 at 10:58 PM, Alexandru Covalschi <568691 at gmail.com>
>>> wrote:
>>>
>>>> Hi list,
>>>>
>>>> I have an issue using tone_detect. The thing is I must detect two very
>>>> short tones - 1400 Hz and 2300 Hz 100ms coming one after another (with 100
>>>> ms pause I suppose) after ~2 seconds the call is answered. I have pretty
>>>> accurate wav file, in audacity I see pikes are 1400 and 2296.
>>>> I've seen that in spandsp sources it is said that accuracy is ~3% so
>>>> that should be enough,
>>>> However tones aren't detected even if I set:
>>>>
>>>> <action application="tone_detect" data="tone_1_0 1395 r 0 set
>>>> tone_first=true 1"/>
>>>> <action application="tone_detect" data="tone_2_0 2295 r 0 set
>>>> tone_second=true 1"/>
>>>> <action application="tone_detect" data="tone_1_1 1396 r 0 set
>>>> tone_first=true 1"/>
>>>> <action application="tone_detect" data="tone_2_1 2296 r 0 set
>>>> tone_second=true 1"/>
>>>> <action application="tone_detect" data="tone_1_2 1397 r 0 set
>>>> tone_first=true 1"/>
>>>> <action application="tone_detect" data="tone_2_2 2297 r 0 set
>>>> tone_second=true 1"/>
>>>> <action application="tone_detect" data="tone_1_3 1398 r 0 set
>>>> tone_first=true 1"/>
>>>> <action application="tone_detect" data="tone_2_3 2298 r 0 set
>>>> tone_second=true 1"/>
>>>> <action application="tone_detect" data="tone_1_4 1399 r 0 set
>>>> tone_first=true 1"/>
>>>> <action application="tone_detect" data="tone_2_4 2299 r 0 set
>>>> tone_second=true 1"/>
>>>> <action application="tone_detect" data="tone_1_5 1400 r 0 set
>>>> tone_first=true 1"/>
>>>> <action application="tone_detect" data="tone_2_5 2300 r 0 set
>>>> tone_second=true 1"/>
>>>> <action application="tone_detect" data="tone_1_6 1401 r 0 set
>>>> tone_first=true 1"/>
>>>> <action application="tone_detect" data="tone_2_6 2301 r 0 set
>>>> tone_second=true 1"/>
>>>> <action application="tone_detect" data="tone_1_7 1402 r 0 set
>>>> tone_first=true 1"/>
>>>> <action application="tone_detect" data="tone_2_7 2302 r 0 set
>>>> tone_second=true 1"/>
>>>>
>>>> Tried to do same using spandsp_start_tone_detect(), but I have no idea
>>>> how to edit spandsp.conf to add a tone matching my needs.
>>>>
>>>> Can someone give me any clue? Any parameter I need to change in sources
>>>> to obtain what I need?
>>>>
>>>> Debian Jessie
>>>> FreeSWITCH version: 1.6.6-13-d2d0b32~64bit (-13-d2d0b32 64bit)
>>>>
>>>>
>>>> Thanks in advance!
>>>> --
>>>> Alexandru Covalschi
>>>> ABRISS-Solutions
>>>> VoIP engineer and system administrator
>>>> tel: +37367398493
>>>> web: http://abriss.solutions/ <http://abs-telecom.com/>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> -
>>>
>>
>>
>> --
>> Alexandru Covalschi
>> ABRISS-Solutions
>> VoIP engineer and system administrator
>> tel: +37367398493
>> web: http://abriss.solutions/ <http://abs-telecom.com/>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From hunterj91 at hotmail.com  Tue Jun 21 20:45:39 2016
From: hunterj91 at hotmail.com (Jonathan Hunter)
Date: Tue, 21 Jun 2016 16:45:39 +0000
Subject: [Freeswitch-users] Call Pickup After a bridge group_call in Dialplan
Message-ID: <DUB112-W99D140DCBCEDBBBBD07A5EBD2B0@phx.gbl>

Hi Guys,
I am routing to a huntgroup using the bridge application and the group_call data, and it all works well.
However I can't get pickup to work, as it wont create the pickup channels.
If I use bridge for a normal call, and add ,pickup, it works fine, its just when bridge and group_call is used.
Am I doing something wrong? Or can some one advise please?
Running Version 1.6
Many thanks
Jon

 		 	   		  
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From nandy1925 at gmail.com  Wed Jun 22 06:25:09 2016
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Wed, 22 Jun 2016 10:25:09 +0800
Subject: [Freeswitch-users] Problem: Installing mod_gsmopen after debian
	package installation
Message-ID: <CAHKqMb49yA6xMFBEs0CHshNDnSkaAdtXe+7NCHXqzMPu9UDmLw@mail.gmail.com>

Hello folks,

I installed FS on Debian (Jessie) package.  Following the GSMopen
installation instruction,  I encountered error in at the

Build and Install Step

There's a "make" error.

mod_gsmopen is built from the source.  Is it required to install FS from
the source, not from deb package,  to properly install gsmopen?

Thanks,
/Nandy
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From nandy1925 at gmail.com  Wed Jun 22 07:07:21 2016
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Wed, 22 Jun 2016 11:07:21 +0800
Subject: [Freeswitch-users] mod_gsmopen make install error
Message-ID: <CAHKqMb7rWh-EU531ik_AFQ0mChCEXkVFZdvd1TUgfTBCC-FmEA@mail.gmail.com>

This is the error message:

make[1]: *** No rule to make target '/usr/src/freeswitch.git/
libfreeswitch.la' needed by 'mod_gsmopen.la'.

I installed libfreeswitch-dev package.  I've done "./configure --prefix=''
" to comply to debian FHS installation tree.

Tks,
/Nandy
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From fanx07 at gmail.com  Wed Jun 22 16:37:19 2016
From: fanx07 at gmail.com (Anonim Stefan)
Date: Wed, 22 Jun 2016 15:37:19 +0300
Subject: [Freeswitch-users] Freeswitch "playback"
In-Reply-To: <CAKbxfG-YobH8c4J-fbDqT1cEB2H2dU8rLp6aMM=FmW8QrvBzhQ@mail.gmail.com>
References: <CAL82Oa0EkKj1QT5mKdrL234xGLTu=SgTSWEfuJE7w0=SFVEWWA@mail.gmail.com>
	<CAF7jiz+yLntmc_dXE1BJNUQOT2Bo+ES5gca2qKN+HcPDJeeuRw@mail.gmail.com>
	<CAKbxfG-YobH8c4J-fbDqT1cEB2H2dU8rLp6aMM=FmW8QrvBzhQ@mail.gmail.com>
Message-ID: <CAO7wsJsKYEuCG6LUh9=WdGYO-TsKCGY22mdoS5-7gOt8ywNtPw@mail.gmail.com>

Thank you,

It works nicely after loading mod_shout for freeswitch 1.4 [1]. I had some
troubles compiling mod_shout for freeswitch 1.6.9, on centos 7 and I ended
up using freeswitch 1.4.

Is there a possibility that the mod_shout will be available in the yum
repos/files for freeswitch 1.6.9?

[1] http://files.freeswitch.org/yum/7/x86_64/

Thanks,
Stefan

On Tue, Jun 14, 2016 at 7:17 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> If this is a stream, mod_httapi is the wrong mod, you would need mod_shout.
>
>
> On Tue, Jun 14, 2016 at 9:53 AM, Benjamin Cropley <
> benjamin.cropley at gmail.com> wrote:
>
>> Did you do wget as the same user FS runs under?
>>
>> Either way, it may be a good idea to do a packet capture (google tcpdump)
>> and check you're 100% making a request for that file.
>>
>> On Tue, Jun 14, 2016 at 1:48 PM, Mititelu Stefan <
>> stefan.mititelu92 at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> I'm trying to do [1] I get [2] even if wget on that link works, on that
>>> machine. It works with local .wav though.
>>>
>>> Do I need to do something else in addition? Do you have some links that
>>> work for you?
>>>
>>> Thanks,
>>> Stefan
>>>
>>>
>>> [1] <action application="playback" data="
>>> http://bbcmedia.ic.llnwd.net/stream/bbcmedia_radio2_mf_p"/>
>>> [2] 2016-06-14 12:11:58.430013 [ERR] mod_httapi.c:2793 File at url [
>>> http://bbcmedia.ic.llnwd.net/stream/bbcmedia_radio2_mf_p] is
>>> unreachable!
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fanx07 at gmail.com  Wed Jun 22 16:39:54 2016
From: fanx07 at gmail.com (Anonim Stefan)
Date: Wed, 22 Jun 2016 15:39:54 +0300
Subject: [Freeswitch-users] Delete previously appended header for the
 bridged leg
In-Reply-To: <CAHtxdDeXjx30gvt+C63vQad+L5jNLZ4OP2nCtcSwfyCD=we=bw@mail.gmail.com>
References: <CAO7wsJsBWBV43qvEs_TBJtX654ZC3dOizXR6osa1e8+oKOZYAg@mail.gmail.com>
	<CAHtxdDeXjx30gvt+C63vQad+L5jNLZ4OP2nCtcSwfyCD=we=bw@mail.gmail.com>
Message-ID: <CAO7wsJsRFGHYXEc58CiVHkvOD6JNgkd1QL=ABzE1_BmBmhVXhA@mail.gmail.com>

Thank you for the answer,

Also, as an alternative, I've been transfering the bridged leg to a
dialplan extension and I was able to unset it from there, as described in
the above tutorials.

Stefan

On Tue, Jun 21, 2016 at 7:14 AM, Sergey Safarov <s.safarov at gmail.com> wrote:

> Try
> originate{origination_uuid=bla}{sip_h_X-HEADER1=VALUE1}sofia/gateway/gateway1/to_uri1
> &bridge({sip_h_X-HEADER1=_undef_,sip_h_X-HEADER2=
> VALUE2}sofia/gateway/gateway1/to_uri2)
>
> ??, 20 ???. 2016 ?. ? 16:16, Anonim Stefan <fanx07 at gmail.com>:
>
>> Hi,
>>
>> I issue this command via python ESL:
>> "
>> originate{origination_uuid=bla}{sip_h_X-HEADER1=VALUE1}sofia/gateway/gateway1/to_uri1
>> &bridge({sip_h_X-HEADER2=VALUE2}sofia/gateway/gateway1/to_uri2)
>> "
>>
>> The problem I have is that HEADER1 is still existent in the bridged leg.
>> I have tried adding &unset(sip_h_X-HEADER1) without success.
>>
>> Also I've tried with [] as said here [1], without succes.
>>
>> What would be the solution to remove the sip_h_X-HEADER1 for the bridged
>> party?
>>
>> [1]
>> https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-ChannelVariablesinDialstrings
>>
>> Thank you,
>> Stefan
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fanx07 at gmail.com  Wed Jun 22 16:50:07 2016
From: fanx07 at gmail.com (Anonim Stefan)
Date: Wed, 22 Jun 2016 15:50:07 +0300
Subject: [Freeswitch-users] OpenBSC <-> Freeswitch ?!
Message-ID: <CAO7wsJv1j4JJkHKoAfm4NQgSD5_4h4r+bSz1bAQRq7e4t3dL4g@mail.gmail.com>

Hi,

Currently, is there such a module on Freeswitch to allow connection to
OpenBSC.

Basically what I want to do is what asterik does in [1].

*LCR = Linux Call Router

[1] http://openbsc.osmocom.org/trac/wiki/OpenBSC_LCR

Thanks,
Stefan
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From ak at hejdu.dk  Wed Jun 22 18:18:37 2016
From: ak at hejdu.dk (Allan Kristensen)
Date: Wed, 22 Jun 2016 16:18:37 +0200
Subject: [Freeswitch-users] Does Freeswtich need rtcp attribute in sdp
 to enable rtcp?
In-Reply-To: <B26B85E5-30CF-4BF0-8E21-D27B1CEE6B9D@jerris.com>
References: <CAA4dF91ygkf6ZDMFO6Z3Se6gtpPa30LRnx5KU6=FYCMPz6Q_oA@mail.gmail.com>
	<B26B85E5-30CF-4BF0-8E21-D27B1CEE6B9D@jerris.com>
Message-ID: <CAA4dF91ZmARuWwPegqdSYAVFPpfaXU47fv9u0YOw9+rzSDgOJQ@mail.gmail.com>

Hello Michael,

Thank you very much, that seems to have solved the problem.

However when running FS in proxy mode (inbound-proxy-media=true) it still
doesn't listen on the rtcp port. Would this be desired behavior?

Best regards,
/Allan

On Fri, Jun 17, 2016 at 3:44 PM, Michael Jerris <mike at jerris.com> wrote:

> upgrade to latest release.
>
> On Jun 17, 2016, at 9:16 AM, Allan Kristensen <ak at hejdu.dk> wrote:
>
> Hello,
>
> Does Freeswitch need the rtcp attribute in the sdp to enable rtcp?
>
> When connecting to asterisk (which does not include a rtcp attribute),
> Freeswitch does not start to listen on the rtcp port (rtp port +1).
>
> When connecting to other Freeswitch servers, which have this attribute in
> the sdp everything works fine.
>
> According to the RFC 3605 it should not be needed, unless it doesn't
> follow the parity rule (rtp port +1).
>
> ...Or is it just me?
>
> Best regards,
>   Allan
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Wed Jun 22 19:07:08 2016
From: mike at jerris.com (Michael Jerris)
Date: Wed, 22 Jun 2016 11:07:08 -0400
Subject: [Freeswitch-users] Freeswitch "playback"
In-Reply-To: <CAO7wsJsKYEuCG6LUh9=WdGYO-TsKCGY22mdoS5-7gOt8ywNtPw@mail.gmail.com>
References: <CAL82Oa0EkKj1QT5mKdrL234xGLTu=SgTSWEfuJE7w0=SFVEWWA@mail.gmail.com>
	<CAF7jiz+yLntmc_dXE1BJNUQOT2Bo+ES5gca2qKN+HcPDJeeuRw@mail.gmail.com>
	<CAKbxfG-YobH8c4J-fbDqT1cEB2H2dU8rLp6aMM=FmW8QrvBzhQ@mail.gmail.com>
	<CAO7wsJsKYEuCG6LUh9=WdGYO-TsKCGY22mdoS5-7gOt8ywNtPw@mail.gmail.com>
Message-ID: <E31C571C-3F2F-48BB-BF30-B6FF1D0664A9@jerris.com>

centos is missing some required deps for mod_shout.  If you can get centos to include the required mp3 libraries, we can add the module to our repos.


> On Jun 22, 2016, at 8:37 AM, Anonim Stefan <fanx07 at gmail.com> wrote:
> 
> Thank you,
> 
> It works nicely after loading mod_shout for freeswitch 1.4 [1]. I had some troubles compiling mod_shout for freeswitch 1.6.9, on centos 7 and I ended up using freeswitch 1.4.
> 
> Is there a possibility that the mod_shout will be available in the yum repos/files for freeswitch 1.6.9?
> 
> [1] http://files.freeswitch.org/yum/7/x86_64/ <http://files.freeswitch.org/yum/7/x86_64/>
> 
> Thanks,
> Stefan
> 
> On Tue, Jun 14, 2016 at 7:17 PM, Anthony Minessale <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
> If this is a stream, mod_httapi is the wrong mod, you would need mod_shout.
> 
> 
> On Tue, Jun 14, 2016 at 9:53 AM, Benjamin Cropley <benjamin.cropley at gmail.com <mailto:benjamin.cropley at gmail.com>> wrote:
> Did you do wget as the same user FS runs under?
> 
> Either way, it may be a good idea to do a packet capture (google tcpdump) and check you're 100% making a request for that file.
> 
> On Tue, Jun 14, 2016 at 1:48 PM, Mititelu Stefan <stefan.mititelu92 at gmail.com <mailto:stefan.mititelu92 at gmail.com>> wrote:
> Hi,
> 
> I'm trying to do [1] I get [2] even if wget on that link works, on that machine. It works with local .wav though.
> 
> Do I need to do something else in addition? Do you have some links that work for you?
> 
> Thanks,
> Stefan
> 
> 
> [1] <action application="playback" data="http://bbcmedia.ic.llnwd.net/stream/bbcmedia_radio2_mf_p <http://bbcmedia.ic.llnwd.net/stream/bbcmedia_radio2_mf_p>"/>
> [2] 2016-06-14 12:11:58.430013 [ERR] mod_httapi.c:2793 File at url [http://bbcmedia.ic.llnwd.net/stream/bbcmedia_radio2_mf_p <http://bbcmedia.ic.llnwd.net/stream/bbcmedia_radio2_mf_p>] is unreachable!
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
> 
> ? http://freeswitch.org/ <http://freeswitch.org/>  ? http://cluecon.com/ <http://cluecon.com/>  ? http://twitter.com/FreeSWITCH <http://twitter.com/FreeSWITCH>
> ? irc.freenode.net <http://irc.freenode.net/> #freeswitch ? http://freeswitch.org/g+ <http://freeswitch.org/g+>
> 
> ClueCon Weekly Development Call 
> ? sip:888 at conference.freeswitch.org <mailto:sip%3A888 at conference.freeswitch.org>  ? +19193869900 <tel:%2B19193869900> 
> 
> https://www.youtube.com/watch?v=9XXgW34t40s <https://www.youtube.com/watch?v=9XXgW34t40s>
> https://www.youtube.com/watch?v=NLaDpGQuZDA <https://www.youtube.com/watch?v=NLaDpGQuZDA>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Wed Jun 22 19:08:33 2016
From: mike at jerris.com (Michael Jerris)
Date: Wed, 22 Jun 2016 11:08:33 -0400
Subject: [Freeswitch-users] Does Freeswtich need rtcp attribute in sdp
	to enable rtcp?
In-Reply-To: <CAA4dF91ZmARuWwPegqdSYAVFPpfaXU47fv9u0YOw9+rzSDgOJQ@mail.gmail.com>
References: <CAA4dF91ygkf6ZDMFO6Z3Se6gtpPa30LRnx5KU6=FYCMPz6Q_oA@mail.gmail.com>
	<B26B85E5-30CF-4BF0-8E21-D27B1CEE6B9D@jerris.com>
	<CAA4dF91ZmARuWwPegqdSYAVFPpfaXU47fv9u0YOw9+rzSDgOJQ@mail.gmail.com>
Message-ID: <2786C5D2-6B37-4507-A027-667A9140E543@jerris.com>

Thats possibly a bug.  Please file in jira.  Is there a reason you want to use proxy mode?  There are very few scenarios where proxy mode is at all useful.

> On Jun 22, 2016, at 10:18 AM, Allan Kristensen <ak at hejdu.dk> wrote:
> 
> Hello Michael,
> 
> Thank you very much, that seems to have solved the problem.
> 
> However when running FS in proxy mode (inbound-proxy-media=true) it still doesn't listen on the rtcp port. Would this be desired behavior?
> 
> Best regards,
> /Allan
> 
> On Fri, Jun 17, 2016 at 3:44 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> upgrade to latest release.
> 
>> On Jun 17, 2016, at 9:16 AM, Allan Kristensen <ak at hejdu.dk <mailto:ak at hejdu.dk>> wrote:
>> 
>> Hello,
>> 
>> Does Freeswitch need the rtcp attribute in the sdp to enable rtcp?
>> 
>> When connecting to asterisk (which does not include a rtcp attribute), Freeswitch does not start to listen on the rtcp port (rtp port +1).
>> 
>> When connecting to other Freeswitch servers, which have this attribute in the sdp everything works fine.
>> 
>> According to the RFC 3605 it should not be needed, unless it doesn't follow the parity rule (rtp port +1).
>> 
>> ...Or is it just me?
>> 
>> Best regards,
>>   Allan

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From kathleen at freeswitch.org  Wed Jun 22 19:55:43 2016
From: kathleen at freeswitch.org (Kathleen King)
Date: Wed, 22 Jun 2016 08:55:43 -0700
Subject: [Freeswitch-users] =?utf-8?q?Can=E2=80=99t_Miss_Keynotes_at_All_A?=
	=?utf-8?q?bout_the_API?=
Message-ID: <CAOStUPaLAyj1Xz4LERcOxif4N1O_kfaNP1sSkG=ouSNNS6b0wg@mail.gmail.com>

[image: All about the API header]




*CAN'T MISS KEYNOTE PRESENTATIONS BY:Oracle, Microsoft, Cisco, IBM, Nexmo,
Zenly, TIBCO, NVIDIA & Khronos Group
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About the API - just check out the keynote lineup
<http://www.allabouttheapi.com/keynote-speakers.aspx> this year. The
keynote speakers will dive into API development, implementation, disruption
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*PLUS - Receive 20% Off with code CLUECON!
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   -

   To register, visit All About The API
   <https://www.tmcnet.com/scripts/events/registration.aspx?theplan=API0716-A&sc=CLUECON>
or
   contact Frank Coppola <fcoppola at tmcnet.com> at 203-852-6800 x131.
   -

   To become an exhibitor and sponsor contact Joe Fabiano
   <jfabiano at tmcnet.com> at 203-852-6800 x132
   or Maureen Gambino <mgambino at tmcnet.com> at 203-852-6800 x109.


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[image: Juan Carlos Soto]
Juan Carlos Soto
Vice President of Hybrid Cloud Integration and API Economy


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Co-Founder & COO


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From ak at hejdu.dk  Wed Jun 22 21:25:30 2016
From: ak at hejdu.dk (Allan Kristensen)
Date: Wed, 22 Jun 2016 19:25:30 +0200
Subject: [Freeswitch-users] Does Freeswtich need rtcp attribute in sdp
 to enable rtcp?
In-Reply-To: <2786C5D2-6B37-4507-A027-667A9140E543@jerris.com>
References: <CAA4dF91ygkf6ZDMFO6Z3Se6gtpPa30LRnx5KU6=FYCMPz6Q_oA@mail.gmail.com>
	<B26B85E5-30CF-4BF0-8E21-D27B1CEE6B9D@jerris.com>
	<CAA4dF91ZmARuWwPegqdSYAVFPpfaXU47fv9u0YOw9+rzSDgOJQ@mail.gmail.com>
	<2786C5D2-6B37-4507-A027-667A9140E543@jerris.com>
Message-ID: <CAA4dF90hpW7SJ2aezpH2024L3t0oWKrNetL6dCG+oYtFO+t_fg@mail.gmail.com>

Hello Michael,

The only reason is performance benefits, but i'm a little uncertain how
much though.

/Allan

On Wed, Jun 22, 2016 at 5:08 PM, Michael Jerris <mike at jerris.com> wrote:

> Thats possibly a bug.  Please file in jira.  Is there a reason you want to
> use proxy mode?  There are very few scenarios where proxy mode is at all
> useful.
>
> On Jun 22, 2016, at 10:18 AM, Allan Kristensen <ak at hejdu.dk> wrote:
>
> Hello Michael,
>
> Thank you very much, that seems to have solved the problem.
>
> However when running FS in proxy mode (inbound-proxy-media=true) it still
> doesn't listen on the rtcp port. Would this be desired behavior?
>
> Best regards,
> /Allan
>
> On Fri, Jun 17, 2016 at 3:44 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> upgrade to latest release.
>>
>> On Jun 17, 2016, at 9:16 AM, Allan Kristensen <ak at hejdu.dk> wrote:
>>
>> Hello,
>>
>> Does Freeswitch need the rtcp attribute in the sdp to enable rtcp?
>>
>> When connecting to asterisk (which does not include a rtcp attribute),
>> Freeswitch does not start to listen on the rtcp port (rtp port +1).
>>
>> When connecting to other Freeswitch servers, which have this attribute in
>> the sdp everything works fine.
>>
>> According to the RFC 3605 it should not be needed, unless it doesn't
>> follow the parity rule (rtp port +1).
>>
>> ...Or is it just me?
>>
>> Best regards,
>>   Allan
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Wed Jun 22 21:32:50 2016
From: mike at jerris.com (Michael Jerris)
Date: Wed, 22 Jun 2016 13:32:50 -0400
Subject: [Freeswitch-users] Does Freeswtich need rtcp attribute in sdp
	to enable rtcp?
In-Reply-To: <CAA4dF90hpW7SJ2aezpH2024L3t0oWKrNetL6dCG+oYtFO+t_fg@mail.gmail.com>
References: <CAA4dF91ygkf6ZDMFO6Z3Se6gtpPa30LRnx5KU6=FYCMPz6Q_oA@mail.gmail.com>
	<B26B85E5-30CF-4BF0-8E21-D27B1CEE6B9D@jerris.com>
	<CAA4dF91ZmARuWwPegqdSYAVFPpfaXU47fv9u0YOw9+rzSDgOJQ@mail.gmail.com>
	<2786C5D2-6B37-4507-A027-667A9140E543@jerris.com>
	<CAA4dF90hpW7SJ2aezpH2024L3t0oWKrNetL6dCG+oYtFO+t_fg@mail.gmail.com>
Message-ID: <6D430370-4956-4BD7-99CD-1AA378F55DBE@jerris.com>

If your only reason is for performance, than proxy mode is not what you want, just use regular mode and late negotiation and avoid transcoding and you will have no performance benefit to proxy mode.

> On Jun 22, 2016, at 1:25 PM, Allan Kristensen <ak at hejdu.dk> wrote:
> 
> Hello Michael,
> 
> The only reason is performance benefits, but i'm a little uncertain how much though.
> 
> /Allan
> 
> On Wed, Jun 22, 2016 at 5:08 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> Thats possibly a bug.  Please file in jira.  Is there a reason you want to use proxy mode?  There are very few scenarios where proxy mode is at all useful.
> 
>> On Jun 22, 2016, at 10:18 AM, Allan Kristensen <ak at hejdu.dk <mailto:ak at hejdu.dk>> wrote:
>> 
>> Hello Michael,
>> 
>> Thank you very much, that seems to have solved the problem.
>> 
>> However when running FS in proxy mode (inbound-proxy-media=true) it still doesn't listen on the rtcp port. Would this be desired behavior?
>> 
>> Best regards,
>> /Allan
>> 
>> On Fri, Jun 17, 2016 at 3:44 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>> upgrade to latest release.
>> 
>>> On Jun 17, 2016, at 9:16 AM, Allan Kristensen <ak at hejdu.dk <mailto:ak at hejdu.dk>> wrote:
>>> 
>>> Hello,
>>> 
>>> Does Freeswitch need the rtcp attribute in the sdp to enable rtcp?
>>> 
>>> When connecting to asterisk (which does not include a rtcp attribute), Freeswitch does not start to listen on the rtcp port (rtp port +1).
>>> 
>>> When connecting to other Freeswitch servers, which have this attribute in the sdp everything works fine.
>>> 
>>> According to the RFC 3605 it should not be needed, unless it doesn't follow the parity rule (rtp port +1).
>>> 
>>> ...Or is it just me?
>>> 
>>> Best regards,
>>>   Allan
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From dragic.dusan at gmail.com  Wed Jun 22 22:46:00 2016
From: dragic.dusan at gmail.com (=?UTF-8?B?RHXFoWFuIERyYWdpxIc=?=)
Date: Wed, 22 Jun 2016 20:46:00 +0200
Subject: [Freeswitch-users] OpenBSC <-> Freeswitch ?!
In-Reply-To: <CAO7wsJv1j4JJkHKoAfm4NQgSD5_4h4r+bSz1bAQRq7e4t3dL4g@mail.gmail.com>
References: <CAO7wsJv1j4JJkHKoAfm4NQgSD5_4h4r+bSz1bAQRq7e4t3dL4g@mail.gmail.com>
Message-ID: <CAPHNFcrBHAreTOtEhC3KN26sVLCre_L77J7MjwerBjSXZmxNog@mail.gmail.com>

IIRC Linux Call Router has a SIP channel driver so there's no need for
a freeswitch LCR endpoint.
Google around a bit and search the openbsc mailing list, you should be
able to find some info how to set it all up.

On 22 June 2016 at 14:50, Anonim Stefan <fanx07 at gmail.com> wrote:
> Hi,
>
> Currently, is there such a module on Freeswitch to allow connection to
> OpenBSC.
>
> Basically what I want to do is what asterik does in [1].
>
> *LCR = Linux Call Router
>
> [1] http://openbsc.osmocom.org/trac/wiki/OpenBSC_LCR
>
> Thanks,
> Stefan
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Du?an Dragi?


From stefan.mititelu92 at gmail.com  Wed Jun 22 23:33:07 2016
From: stefan.mititelu92 at gmail.com (Mititelu Stefan)
Date: Wed, 22 Jun 2016 22:33:07 +0300
Subject: [Freeswitch-users] OpenBSC <-> Freeswitch ?!
In-Reply-To: <CAPHNFcrBHAreTOtEhC3KN26sVLCre_L77J7MjwerBjSXZmxNog@mail.gmail.com>
References: <CAO7wsJv1j4JJkHKoAfm4NQgSD5_4h4r+bSz1bAQRq7e4t3dL4g@mail.gmail.com>
	<CAPHNFcrBHAreTOtEhC3KN26sVLCre_L77J7MjwerBjSXZmxNog@mail.gmail.com>
Message-ID: <CAL82Oa2U1Wv0ycYM=tE9Ts7ZZ-hao3X2K7UZ8fb=W612OoGRRg@mail.gmail.com>

Thank you.

Stefan

On Wed, Jun 22, 2016 at 9:46 PM, Du?an Dragi? <dragic.dusan at gmail.com>
wrote:

> IIRC Linux Call Router has a SIP channel driver so there's no need for
> a freeswitch LCR endpoint.
> Google around a bit and search the openbsc mailing list, you should be
> able to find some info how to set it all up.
>
> On 22 June 2016 at 14:50, Anonim Stefan <fanx07 at gmail.com> wrote:
> > Hi,
> >
> > Currently, is there such a module on Freeswitch to allow connection to
> > OpenBSC.
> >
> > Basically what I want to do is what asterik does in [1].
> >
> > *LCR = Linux Call Router
> >
> > [1] http://openbsc.osmocom.org/trac/wiki/OpenBSC_LCR
> >
> > Thanks,
> > Stefan
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> --
> Du?an Dragi?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From ak at hejdu.dk  Thu Jun 23 15:57:05 2016
From: ak at hejdu.dk (Allan Kristensen)
Date: Thu, 23 Jun 2016 13:57:05 +0200
Subject: [Freeswitch-users] Does Freeswtich need rtcp attribute in sdp
 to enable rtcp?
In-Reply-To: <6D430370-4956-4BD7-99CD-1AA378F55DBE@jerris.com>
References: <CAA4dF91ygkf6ZDMFO6Z3Se6gtpPa30LRnx5KU6=FYCMPz6Q_oA@mail.gmail.com>
	<B26B85E5-30CF-4BF0-8E21-D27B1CEE6B9D@jerris.com>
	<CAA4dF91ZmARuWwPegqdSYAVFPpfaXU47fv9u0YOw9+rzSDgOJQ@mail.gmail.com>
	<2786C5D2-6B37-4507-A027-667A9140E543@jerris.com>
	<CAA4dF90hpW7SJ2aezpH2024L3t0oWKrNetL6dCG+oYtFO+t_fg@mail.gmail.com>
	<6D430370-4956-4BD7-99CD-1AA378F55DBE@jerris.com>
Message-ID: <CAA4dF90eqCBpovR2rwkpTsmAq2AExUC5dhVYDR5M42EF1M=r=g@mail.gmail.com>

Sounds good, i'll keep with regular mode and get working rtcp stats...

Thanks..
 Allan


On Wed, Jun 22, 2016 at 7:32 PM, Michael Jerris <mike at jerris.com> wrote:

> If your only reason is for performance, than proxy mode is not what you
> want, just use regular mode and late negotiation and avoid transcoding and
> you will have no performance benefit to proxy mode.
>
> On Jun 22, 2016, at 1:25 PM, Allan Kristensen <ak at hejdu.dk> wrote:
>
> Hello Michael,
>
> The only reason is performance benefits, but i'm a little uncertain how
> much though.
>
> /Allan
>
> On Wed, Jun 22, 2016 at 5:08 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> Thats possibly a bug.  Please file in jira.  Is there a reason you want
>> to use proxy mode?  There are very few scenarios where proxy mode is at all
>> useful.
>>
>> On Jun 22, 2016, at 10:18 AM, Allan Kristensen <ak at hejdu.dk> wrote:
>>
>> Hello Michael,
>>
>> Thank you very much, that seems to have solved the problem.
>>
>> However when running FS in proxy mode (inbound-proxy-media=true) it still
>> doesn't listen on the rtcp port. Would this be desired behavior?
>>
>> Best regards,
>> /Allan
>>
>> On Fri, Jun 17, 2016 at 3:44 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> upgrade to latest release.
>>>
>>> On Jun 17, 2016, at 9:16 AM, Allan Kristensen <ak at hejdu.dk> wrote:
>>>
>>> Hello,
>>>
>>> Does Freeswitch need the rtcp attribute in the sdp to enable rtcp?
>>>
>>> When connecting to asterisk (which does not include a rtcp attribute),
>>> Freeswitch does not start to listen on the rtcp port (rtp port +1).
>>>
>>> When connecting to other Freeswitch servers, which have this attribute
>>> in the sdp everything works fine.
>>>
>>> According to the RFC 3605 it should not be needed, unless it doesn't
>>> follow the parity rule (rtp port +1).
>>>
>>> ...Or is it just me?
>>>
>>> Best regards,
>>>   Allan
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ashwinrkjain at gmail.com  Thu Jun 23 17:56:41 2016
From: ashwinrkjain at gmail.com (Ashwin Jain)
Date: Thu, 23 Jun 2016 19:26:41 +0530
Subject: [Freeswitch-users] Audio skip issue with simultaneous ringing
In-Reply-To: <CAM9SNVSQSNp7gTfCV2gJ6t8BYmVeKFCP01dC9WN+b0-RENMMAg@mail.gmail.com>
References: <CAM9SNVTQJo5jCZ+jb9Yy6fErb7UdUtH1QpbNn=8kPkdaP=m0vg@mail.gmail.com>
	<CAEwTsAjozSr2eQyPht1nxP5AophRUeqgxYaWtqBrn5+HA9o1zw@mail.gmail.com>
	<CAM9SNVSQSNp7gTfCV2gJ6t8BYmVeKFCP01dC9WN+b0-RENMMAg@mail.gmail.com>
Message-ID: <CAM9SNVTXTNit_MwrDsT1TZPDeNoY=rEN=2rqhjEKNSkqjQADJg@mail.gmail.com>

Hi Michael,

Can you please confirm that it is the freeswitch issue and not our
configuration issue? That would be of really great help.

On Tue, Jun 21, 2016 at 1:46 AM, Ashwin Jain <ashwinrkjain at gmail.com> wrote:

> Sure. We can update to the latest version. Can you confirm if this is the
> issue, or is it some setup/configuration issue with our system?
>
> On Mon, Jun 20, 2016 at 9:04 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> You are using older versions, I would first check if we have fixed this
>> by testing the latest releases.  1.4 is only receiving critical security
>> releases now and would not receive a fix if we found a problem.
>>
>> On Saturday, June 18, 2016, Ashwin Jain <ashwinrkjain at gmail.com> wrote:
>>
>>> Hi all,
>>>
>>> We are running Freeswitch version 1.4.26. We are facing a major problem
>>> related to audio skip with simultaneous ringing. In one flow we are ringing
>>> around 5 agents at the same time. Whenever agent 1 (any one of the agent)
>>> picks up the call, other legs (to other agents) gets a hangup event. Due to
>>> this, hangup event to all the other agent "CALL_ANSWER" event on the other
>>> side gets delayed by around 2-3 seconds and generally caller missed the
>>> first word from the agent.
>>>
>>> Event 1: Call Answered by agent 1
>>> Event 2: Hangup for agent 2 (due to LOSE_RACE)
>>> Event 3: Hangup for agent 3 (due to LOSE_RACE)
>>> Event 4: Hangup for agent 4 (due to LOSE_RACE)
>>> Event 5: Hangup for agent 5 (due to LOSE_RACE)
>>> Event 6: Call answered to caller
>>>
>>> Time different between Event 1 and Event 6 is about 2-3 seconds.
>>> Any help would be really really helpful
>>>
>>> --
>>> Thanks and Regards,
>>> Ashwin Jain
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Ashwin Jain*
>


-- 

*Ashwin Jain*
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From david.ponzone at gmail.com  Thu Jun 23 18:35:39 2016
From: david.ponzone at gmail.com (David Ponzone)
Date: Thu, 23 Jun 2016 16:35:39 +0200
Subject: [Freeswitch-users] Question regarding UUID stacking in a FIFO
Message-ID: <64F76782-7F99-40AD-9134-7B1A2340B722@gmail.com>

Guys,

I am searching for a way to stack UUID into a FIFO.
What I need is something like db insert/realm/key/${uuid} or hash insert/realm/key/${uuid} but that I need to keep several uuids with the same realm/key tuple.

The ideal would be:
I first insert realm/key/${uuid1}
then insert real/ley/{uuid2}

then if I do select/realm/key, it pops out ${uuid1)
I can then manually remove it, and the next select will pop out ${uuid2}

I don?t think it?s possible with hash/db, so does anyone a way/trick to achieve that ? 

(the goal is to intercept several simultaneous incoming calls in the RINGING state)

Thank you

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr <blocked::http://www.ipeva.fr/>  -   www.ipeva-studio.com <blocked::http://www.ipeva-studio.com/>

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




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From mike at jerris.com  Thu Jun 23 19:26:41 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 23 Jun 2016 11:26:41 -0400
Subject: [Freeswitch-users] Audio skip issue with simultaneous ringing
In-Reply-To: <CAM9SNVTXTNit_MwrDsT1TZPDeNoY=rEN=2rqhjEKNSkqjQADJg@mail.gmail.com>
References: <CAM9SNVTQJo5jCZ+jb9Yy6fErb7UdUtH1QpbNn=8kPkdaP=m0vg@mail.gmail.com>
	<CAEwTsAjozSr2eQyPht1nxP5AophRUeqgxYaWtqBrn5+HA9o1zw@mail.gmail.com>
	<CAM9SNVSQSNp7gTfCV2gJ6t8BYmVeKFCP01dC9WN+b0-RENMMAg@mail.gmail.com>
	<CAM9SNVTXTNit_MwrDsT1TZPDeNoY=rEN=2rqhjEKNSkqjQADJg@mail.gmail.com>
Message-ID: <CAEwTsAg0+rVhJpqiQWePJ5vUv5xN_pyFnc=RGSvE2tcgGjLFFw@mail.gmail.com>

My suggestion is for you to confirm this by testing the latest release.
There have been over 1000 bug fixes since the version you are using, I
don't recall if this specific issue is included in those or not.

On Thursday, June 23, 2016, Ashwin Jain <ashwinrkjain at gmail.com> wrote:

> Hi Michael,
>
> Can you please confirm that it is the freeswitch issue and not our
> configuration issue? That would be of really great help.
>
> On Tue, Jun 21, 2016 at 1:46 AM, Ashwin Jain <ashwinrkjain at gmail.com
> <javascript:_e(%7B%7D,'cvml','ashwinrkjain at gmail.com');>> wrote:
>
>> Sure. We can update to the latest version. Can you confirm if this is the
>> issue, or is it some setup/configuration issue with our system?
>>
>> On Mon, Jun 20, 2016 at 9:04 PM, Michael Jerris <mike at jerris.com
>> <javascript:_e(%7B%7D,'cvml','mike at jerris.com');>> wrote:
>>
>>> You are using older versions, I would first check if we have fixed this
>>> by testing the latest releases.  1.4 is only receiving critical security
>>> releases now and would not receive a fix if we found a problem.
>>>
>>> On Saturday, June 18, 2016, Ashwin Jain <ashwinrkjain at gmail.com
>>> <javascript:_e(%7B%7D,'cvml','ashwinrkjain at gmail.com');>> wrote:
>>>
>>>> Hi all,
>>>>
>>>> We are running Freeswitch version 1.4.26. We are facing a major problem
>>>> related to audio skip with simultaneous ringing. In one flow we are ringing
>>>> around 5 agents at the same time. Whenever agent 1 (any one of the agent)
>>>> picks up the call, other legs (to other agents) gets a hangup event. Due to
>>>> this, hangup event to all the other agent "CALL_ANSWER" event on the other
>>>> side gets delayed by around 2-3 seconds and generally caller missed the
>>>> first word from the agent.
>>>>
>>>> Event 1: Call Answered by agent 1
>>>> Event 2: Hangup for agent 2 (due to LOSE_RACE)
>>>> Event 3: Hangup for agent 3 (due to LOSE_RACE)
>>>> Event 4: Hangup for agent 4 (due to LOSE_RACE)
>>>> Event 5: Hangup for agent 5 (due to LOSE_RACE)
>>>> Event 6: Call answered to caller
>>>>
>>>> Time different between Event 1 and Event 6 is about 2-3 seconds.
>>>> Any help would be really really helpful
>>>>
>>>> --
>>>> Thanks and Regards,
>>>> Ashwin Jain
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> *Ashwin Jain*
>>
>
>
> --
>
> *Ashwin Jain*
>
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From aubalde at presenceco.com  Thu Jun 23 20:20:16 2016
From: aubalde at presenceco.com (=?UTF-8?Q?Agust=C3=AD_Ubalde?=)
Date: Thu, 23 Jun 2016 18:20:16 +0200
Subject: [Freeswitch-users] Changing Auto stun / rtp / DTLS
Message-ID: <CAKThY069UjJHRuu9OL0Pi8CMhP=qR8V97RUirEWKwGWFC_8NkA@mail.gmail.com>

HI all,

In the log file I observed the trace *Changing Auto stun / rtp / DTLS*  It
only happens on connections from a particular IP. Anyone know explain to
that is?


Regards,

*PRESENCE TECHNOLOGY*
*Agust? Ubalde Bellot*
Chief Developer
C/ Comte Urgell 240 3A
Barcelona 08036
aubalde at presenceco.com

Ph: +34 93 10 10 300
Fx: +34 93 10 10 333

*www.presenceco.com* <http://www.presenceco.com/>

*Follow us on:*

*[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
<http://www.youtube.com/PresenceTechnology>*[image: in]*
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<http://www.slideshare.net/PresenceTech>*[image: fb]*
<http://www.facebook.com/presencetech>

For additional information, please visit our website *www.presenceco.com*
<http://www.presenceco.com/>

-- 



*Presence Technology - DisclaimerThis message, its content and any file 
attached thereto is for the intended recipient only and is confidential and 
/or privileged. If you have received this e-mail in error or had access to 
it, you should note that the information in it is private and any use 
thereof is unauthorized. In such an event please notify us by e-mail or by 
telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by 
whatsoever means and any transmission or dissemination thereof to other 
persons is prohibited. It should be deleted immediately from your system. 
Presence Technology reserves the right to take legal action against any 
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has emitted.*

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<http://www.presenceco.com/>*

 
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From danny.gershman at gmail.com  Thu Jun 23 21:12:25 2016
From: danny.gershman at gmail.com (Danny Gershman)
Date: Thu, 23 Jun 2016 17:12:25 +0000
Subject: [Freeswitch-users] conference play video
In-Reply-To: <618BE480-F3CD-4903-85A5-923FD12C2941@jerris.com>
References: <CAEorZymOmq5b88qBgvCEO5-eoucLx99Si-cYu4HNKi4-Su16fg@mail.gmail.com>
	<618BE480-F3CD-4903-85A5-923FD12C2941@jerris.com>
Message-ID: <CAEorZy=VFyVvfx+dCCfuBZvR6sS1tGgBfjnTZ_CUj=vj4NS0cA@mail.gmail.com>

I'm trying on 1.6.9 and I'm still not getting results.  Is there some
specific condition in which this will work.  If I have muxing turned on,
how does that display back to a softphone?  Should this be played in a
specific channel?  Is playing a video file to all users in a conference,
what would the expected response be?

On Thu, Jun 16, 2016 at 8:08 PM Michael Jerris <mike at jerris.com> wrote:

> Try 1.6.9 as thats now released, if thats not working, take a look at the
> debug logs and see what it says.  One common issue is playing mp4 files
> requires the libav-extras package on debian.
>
>
> On Jun 16, 2016, at 6:54 PM, Danny Gershman <danny.gershman at gmail.com>
> wrote:
>
> I'm having an issue playing back video in a conference, running (1.6.8 tag
> and loaded up mod_av), which it only plays audio back (no visual).  I'm
> dialed in using a softphone with VP8 codec.
>
> I run command:
>
> conference <conf> play http://example.org/play.mp4
>
> Any pointers.
>
> Thanks,
> Danny Gershman
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From anthony.minessale at gmail.com  Thu Jun 23 21:16:22 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 23 Jun 2016 12:16:22 -0500
Subject: [Freeswitch-users] Changing Auto stun / rtp / DTLS
In-Reply-To: <CAKThY069UjJHRuu9OL0Pi8CMhP=qR8V97RUirEWKwGWFC_8NkA@mail.gmail.com>
References: <CAKThY069UjJHRuu9OL0Pi8CMhP=qR8V97RUirEWKwGWFC_8NkA@mail.gmail.com>
Message-ID: <CAKbxfG8-6W-hT=cEdn=3QBCic8Os7FY1vadVA-r7m4r-6+yo5A@mail.gmail.com>

It happens when a candidate has not gotten any stun from the expected
places but it gets some from another place that is not the right address
but does have the right credentials.
It will lock on to the remote destination for that candidate instead.  This
is usually encountered when the other side is asymmetric nat.

On Thu, Jun 23, 2016 at 11:20 AM, Agust? Ubalde <aubalde at presenceco.com>
wrote:

> HI all,
>
> In the log file I observed the trace *Changing Auto stun / rtp / DTLS*
> It only happens on connections from a particular IP. Anyone know explain to
> that is?
>
>
> Regards,
>
> *PRESENCE TECHNOLOGY*
> *Agust? Ubalde Bellot*
> Chief Developer
> C/ Comte Urgell 240 3A
> Barcelona 08036
> aubalde at presenceco.com
>
> Ph: +34 93 10 10 300
> Fx: +34 93 10 10 333
>
> *www.presenceco.com* <http://www.presenceco.com/>
>
> *Follow us on:*
>
> *[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
> <http://www.youtube.com/PresenceTechnology>*[image: in]*
> <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]*
> <http://www.slideshare.net/PresenceTech>*[image: fb]*
> <http://www.facebook.com/presencetech>
>
> For additional information, please visit our website *www.presenceco.com*
> <http://www.presenceco.com/>
>
>
> *Presence Technology - DisclaimerThis message, its content and any file
> attached thereto is for the intended recipient only and is confidential and
> /or privileged. If you have received this e-mail in error or had access to
> it, you should note that the information in it is private and any use
> thereof is unauthorized. In such an event please notify us by e-mail or by
> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by
> whatsoever means and any transmission or dissemination thereof to other
> persons is prohibited. It should be deleted immediately from your system.
> Presence Technology reserves the right to take legal action against any
> persons unlawfully gaining access to the content of any external message it
> has emitted.*
>
> *For additional information, please visit our website **www.presenceco.com
> <http://www.presenceco.com/>*
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From anthony.minessale at gmail.com  Thu Jun 23 21:28:45 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 23 Jun 2016 12:28:45 -0500
Subject: [Freeswitch-users] conference play video
In-Reply-To: <CAEorZy=VFyVvfx+dCCfuBZvR6sS1tGgBfjnTZ_CUj=vj4NS0cA@mail.gmail.com>
References: <CAEorZymOmq5b88qBgvCEO5-eoucLx99Si-cYu4HNKi4-Su16fg@mail.gmail.com>
	<618BE480-F3CD-4903-85A5-923FD12C2941@jerris.com>
	<CAEorZy=VFyVvfx+dCCfuBZvR6sS1tGgBfjnTZ_CUj=vj4NS0cA@mail.gmail.com>
Message-ID: <CAKbxfG-_ouQdkOra2vkPgsoOpkPF6BmuHYzfDMN1BnQSmJfBSg@mail.gmail.com>

Like mike said, make sure you have main contrib non-free for you repos and

libavcodec-extra



On Thu, Jun 23, 2016 at 12:12 PM, Danny Gershman <danny.gershman at gmail.com>
wrote:

> I'm trying on 1.6.9 and I'm still not getting results.  Is there some
> specific condition in which this will work.  If I have muxing turned on,
> how does that display back to a softphone?  Should this be played in a
> specific channel?  Is playing a video file to all users in a conference,
> what would the expected response be?
>
> On Thu, Jun 16, 2016 at 8:08 PM Michael Jerris <mike at jerris.com> wrote:
>
>> Try 1.6.9 as thats now released, if thats not working, take a look at the
>> debug logs and see what it says.  One common issue is playing mp4 files
>> requires the libav-extras package on debian.
>>
>>
>> On Jun 16, 2016, at 6:54 PM, Danny Gershman <danny.gershman at gmail.com>
>> wrote:
>>
>> I'm having an issue playing back video in a conference, running (1.6.8
>> tag and loaded up mod_av), which it only plays audio back (no visual).  I'm
>> dialed in using a softphone with VP8 codec.
>>
>> I run command:
>>
>> conference <conf> play http://example.org/play.mp4
>>
>> Any pointers.
>>
>> Thanks,
>> Danny Gershman
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From danny.gershman at gmail.com  Thu Jun 23 21:43:09 2016
From: danny.gershman at gmail.com (Danny Gershman)
Date: Thu, 23 Jun 2016 17:43:09 +0000
Subject: [Freeswitch-users] conference play video
In-Reply-To: <CAKbxfG-_ouQdkOra2vkPgsoOpkPF6BmuHYzfDMN1BnQSmJfBSg@mail.gmail.com>
References: <CAEorZymOmq5b88qBgvCEO5-eoucLx99Si-cYu4HNKi4-Su16fg@mail.gmail.com>
	<618BE480-F3CD-4903-85A5-923FD12C2941@jerris.com>
	<CAEorZy=VFyVvfx+dCCfuBZvR6sS1tGgBfjnTZ_CUj=vj4NS0cA@mail.gmail.com>
	<CAKbxfG-_ouQdkOra2vkPgsoOpkPF6BmuHYzfDMN1BnQSmJfBSg@mail.gmail.com>
Message-ID: <CAEorZykBsfBcS2-MMDdCeJak+ra3Oy2cSjm64Us+txGeyQhadQ@mail.gmail.com>

Already have it there, FS was compiled with it.

On Thu, Jun 23, 2016 at 1:29 PM Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Like mike said, make sure you have main contrib non-free for you repos
> and
>
> libavcodec-extra
>
>
>
> On Thu, Jun 23, 2016 at 12:12 PM, Danny Gershman <danny.gershman at gmail.com
> > wrote:
>
>> I'm trying on 1.6.9 and I'm still not getting results.  Is there some
>> specific condition in which this will work.  If I have muxing turned on,
>> how does that display back to a softphone?  Should this be played in a
>> specific channel?  Is playing a video file to all users in a conference,
>> what would the expected response be?
>>
>> On Thu, Jun 16, 2016 at 8:08 PM Michael Jerris <mike at jerris.com> wrote:
>>
>>> Try 1.6.9 as thats now released, if thats not working, take a look at
>>> the debug logs and see what it says.  One common issue is playing mp4 files
>>> requires the libav-extras package on debian.
>>>
>>>
>>> On Jun 16, 2016, at 6:54 PM, Danny Gershman <danny.gershman at gmail.com>
>>> wrote:
>>>
>>> I'm having an issue playing back video in a conference, running (1.6.8
>>> tag and loaded up mod_av), which it only plays audio back (no visual).  I'm
>>> dialed in using a softphone with VP8 codec.
>>>
>>> I run command:
>>>
>>> conference <conf> play http://example.org/play.mp4
>>>
>>> Any pointers.
>>>
>>> Thanks,
>>> Danny Gershman
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From anthony.minessale at gmail.com  Thu Jun 23 21:51:37 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 23 Jun 2016 12:51:37 -0500
Subject: [Freeswitch-users] conference play video
In-Reply-To: <CAEorZykBsfBcS2-MMDdCeJak+ra3Oy2cSjm64Us+txGeyQhadQ@mail.gmail.com>
References: <CAEorZymOmq5b88qBgvCEO5-eoucLx99Si-cYu4HNKi4-Su16fg@mail.gmail.com>
	<618BE480-F3CD-4903-85A5-923FD12C2941@jerris.com>
	<CAEorZy=VFyVvfx+dCCfuBZvR6sS1tGgBfjnTZ_CUj=vj4NS0cA@mail.gmail.com>
	<CAKbxfG-_ouQdkOra2vkPgsoOpkPF6BmuHYzfDMN1BnQSmJfBSg@mail.gmail.com>
	<CAEorZykBsfBcS2-MMDdCeJak+ra3Oy2cSjm64Us+txGeyQhadQ@mail.gmail.com>
Message-ID: <CAKbxfG9Vd-75Sp=yZ3EiXZLwyoZH-fs=C9jvVHZCRPotmcEnrA@mail.gmail.com>

You need to check your environment again because if you have it setup
properly it will function.
Check your mod_av settings and versions of libav as pulled in from
freeswitch-video-deps-most


On Thu, Jun 23, 2016 at 12:43 PM, Danny Gershman <danny.gershman at gmail.com>
wrote:

> Already have it there, FS was compiled with it.
>
> On Thu, Jun 23, 2016 at 1:29 PM Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Like mike said, make sure you have main contrib non-free for you repos
>> and
>>
>> libavcodec-extra
>>
>>
>>
>> On Thu, Jun 23, 2016 at 12:12 PM, Danny Gershman <
>> danny.gershman at gmail.com> wrote:
>>
>>> I'm trying on 1.6.9 and I'm still not getting results.  Is there some
>>> specific condition in which this will work.  If I have muxing turned on,
>>> how does that display back to a softphone?  Should this be played in a
>>> specific channel?  Is playing a video file to all users in a conference,
>>> what would the expected response be?
>>>
>>> On Thu, Jun 16, 2016 at 8:08 PM Michael Jerris <mike at jerris.com> wrote:
>>>
>>>> Try 1.6.9 as thats now released, if thats not working, take a look at
>>>> the debug logs and see what it says.  One common issue is playing mp4 files
>>>> requires the libav-extras package on debian.
>>>>
>>>>
>>>> On Jun 16, 2016, at 6:54 PM, Danny Gershman <danny.gershman at gmail.com>
>>>> wrote:
>>>>
>>>> I'm having an issue playing back video in a conference, running (1.6.8
>>>> tag and loaded up mod_av), which it only plays audio back (no visual).  I'm
>>>> dialed in using a softphone with VP8 codec.
>>>>
>>>> I run command:
>>>>
>>>> conference <conf> play http://example.org/play.mp4
>>>>
>>>> Any pointers.
>>>>
>>>> Thanks,
>>>> Danny Gershman
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>>
>> ? http://freeswitch.org/  ? http://cluecon.com/  ?
>> http://twitter.com/FreeSWITCH
>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
>> <http://freeswitch.org/g+>*
>>
>> ClueCon Weekly Development Call
>> ? sip:888 at conference.freeswitch.org  ? +19193869900
>>
>> https://www.youtube.com/watch?v=9XXgW34t40s
>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
-------------- next part --------------
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From mike at jerris.com  Thu Jun 23 21:53:48 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 23 Jun 2016 13:53:48 -0400
Subject: [Freeswitch-users] conference play video
In-Reply-To: <CAEorZykBsfBcS2-MMDdCeJak+ra3Oy2cSjm64Us+txGeyQhadQ@mail.gmail.com>
References: <CAEorZymOmq5b88qBgvCEO5-eoucLx99Si-cYu4HNKi4-Su16fg@mail.gmail.com>
	<618BE480-F3CD-4903-85A5-923FD12C2941@jerris.com>
	<CAEorZy=VFyVvfx+dCCfuBZvR6sS1tGgBfjnTZ_CUj=vj4NS0cA@mail.gmail.com>
	<CAKbxfG-_ouQdkOra2vkPgsoOpkPF6BmuHYzfDMN1BnQSmJfBSg@mail.gmail.com>
	<CAEorZykBsfBcS2-MMDdCeJak+ra3Oy2cSjm64Us+txGeyQhadQ@mail.gmail.com>
Message-ID: <B319A4CC-3F70-4B58-93B2-556A291F3D27@jerris.com>

I would confirm for sure thats installed, then look at the debug and confirm its actually negotiating a video codec.  Its really hard to make guesses with no information to go on so we are just making wild guesses here.

> On Jun 23, 2016, at 1:43 PM, Danny Gershman <danny.gershman at gmail.com> wrote:
> 
> Already have it there, FS was compiled with it.
> 
> On Thu, Jun 23, 2016 at 1:29 PM Anthony Minessale <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
> Like mike said, make sure you have main contrib non-free for you repos and 
> libavcodec-extra
> 
> 
> 
> 
> On Thu, Jun 23, 2016 at 12:12 PM, Danny Gershman <danny.gershman at gmail.com <mailto:danny.gershman at gmail.com>> wrote:
> I'm trying on 1.6.9 and I'm still not getting results.  Is there some specific condition in which this will work.  If I have muxing turned on, how does that display back to a softphone?  Should this be played in a specific channel?  Is playing a video file to all users in a conference, what would the expected response be?
> 
> On Thu, Jun 16, 2016 at 8:08 PM Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> Try 1.6.9 as thats now released, if thats not working, take a look at the debug logs and see what it says.  One common issue is playing mp4 files requires the libav-extras package on debian.
> 
> 
> 
>> On Jun 16, 2016, at 6:54 PM, Danny Gershman <danny.gershman at gmail.com <mailto:danny.gershman at gmail.com>> wrote:
>> 
> 
>> I'm having an issue playing back video in a conference, running (1.6.8 tag and loaded up mod_av), which it only plays audio back (no visual).  I'm dialed in using a softphone with VP8 codec.
>> 
>> I run command:
>> 
>> conference <conf> play http://example.org/play.mp4 <http://example.org/play.mp4>
>> 
>> Any pointers.
>> 
>> Thanks,
>> Danny Gershman
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From danny.gershman at gmail.com  Thu Jun 23 21:53:53 2016
From: danny.gershman at gmail.com (Danny Gershman)
Date: Thu, 23 Jun 2016 17:53:53 +0000
Subject: [Freeswitch-users] conference play video
In-Reply-To: <CAKbxfG9Vd-75Sp=yZ3EiXZLwyoZH-fs=C9jvVHZCRPotmcEnrA@mail.gmail.com>
References: <CAEorZymOmq5b88qBgvCEO5-eoucLx99Si-cYu4HNKi4-Su16fg@mail.gmail.com>
	<618BE480-F3CD-4903-85A5-923FD12C2941@jerris.com>
	<CAEorZy=VFyVvfx+dCCfuBZvR6sS1tGgBfjnTZ_CUj=vj4NS0cA@mail.gmail.com>
	<CAKbxfG-_ouQdkOra2vkPgsoOpkPF6BmuHYzfDMN1BnQSmJfBSg@mail.gmail.com>
	<CAEorZykBsfBcS2-MMDdCeJak+ra3Oy2cSjm64Us+txGeyQhadQ@mail.gmail.com>
	<CAKbxfG9Vd-75Sp=yZ3EiXZLwyoZH-fs=C9jvVHZCRPotmcEnrA@mail.gmail.com>
Message-ID: <CAEorZy=AtQ1zf4mUUK7fmpzUoznO=6TG7LLX2Gq_7ec=_EYLpw@mail.gmail.com>

What would I expect to see?  When video is muxed, each display is a
channel.  If I do a conference play, what happens, how does this translate
in a muxed display?

On Thu, Jun 23, 2016 at 1:52 PM Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> You need to check your environment again because if you have it setup
> properly it will function.
> Check your mod_av settings and versions of libav as pulled in from
> freeswitch-video-deps-most
>
>
> On Thu, Jun 23, 2016 at 12:43 PM, Danny Gershman <danny.gershman at gmail.com
> > wrote:
>
>> Already have it there, FS was compiled with it.
>>
>> On Thu, Jun 23, 2016 at 1:29 PM Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> Like mike said, make sure you have main contrib non-free for you repos
>>> and
>>>
>>> libavcodec-extra
>>>
>>>
>>>
>>> On Thu, Jun 23, 2016 at 12:12 PM, Danny Gershman <
>>> danny.gershman at gmail.com> wrote:
>>>
>>>> I'm trying on 1.6.9 and I'm still not getting results.  Is there some
>>>> specific condition in which this will work.  If I have muxing turned on,
>>>> how does that display back to a softphone?  Should this be played in a
>>>> specific channel?  Is playing a video file to all users in a conference,
>>>> what would the expected response be?
>>>>
>>>> On Thu, Jun 16, 2016 at 8:08 PM Michael Jerris <mike at jerris.com> wrote:
>>>>
>>>>> Try 1.6.9 as thats now released, if thats not working, take a look at
>>>>> the debug logs and see what it says.  One common issue is playing mp4 files
>>>>> requires the libav-extras package on debian.
>>>>>
>>>>>
>>>>> On Jun 16, 2016, at 6:54 PM, Danny Gershman <danny.gershman at gmail.com>
>>>>> wrote:
>>>>>
>>>>> I'm having an issue playing back video in a conference, running (1.6.8
>>>>> tag and loaded up mod_av), which it only plays audio back (no visual).  I'm
>>>>> dialed in using a softphone with VP8 codec.
>>>>>
>>>>> I run command:
>>>>>
>>>>> conference <conf> play http://example.org/play.mp4
>>>>>
>>>>> Any pointers.
>>>>>
>>>>> Thanks,
>>>>> Danny Gershman
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>>>
>>> ? http://freeswitch.org/  ? http://cluecon.com/  ?
>>> http://twitter.com/FreeSWITCH
>>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
>>> <http://freeswitch.org/g+>*
>>>
>>> ClueCon Weekly Development Call
>>> ? sip:888 at conference.freeswitch.org  ? +19193869900
>>>
>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From jmoran at secureachsystems.com  Thu Jun 23 21:54:54 2016
From: jmoran at secureachsystems.com (Jason Moran)
Date: Thu, 23 Jun 2016 13:54:54 -0400
Subject: [Freeswitch-users] Debian 8 package i386?
Message-ID: <04e201d1cd78$5b890110$129b0330$@secureachsystems.com>

I have VMs of Deb8 for both 64bit and 32bit/i386. 
The instructions work great for 64bit, and FS installed as expected.

 

However, on i386 the command

# apt-get update && apt-get install -y freeswitch-meta-all

returns

E: Unable to locate package freeswitch-meta-all

 

I looked in
http://files.freeswitch.org/repo/deb/freeswitch-1.6/db/packages.db and see
i386 packages for 1.4 (Wheezy) but for 1.6 I only see amd64.

Is our old x86 server unsupported now?

 

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From danny.gershman at gmail.com  Thu Jun 23 21:55:50 2016
From: danny.gershman at gmail.com (Danny Gershman)
Date: Thu, 23 Jun 2016 17:55:50 +0000
Subject: [Freeswitch-users] conference play video
In-Reply-To: <B319A4CC-3F70-4B58-93B2-556A291F3D27@jerris.com>
References: <CAEorZymOmq5b88qBgvCEO5-eoucLx99Si-cYu4HNKi4-Su16fg@mail.gmail.com>
	<618BE480-F3CD-4903-85A5-923FD12C2941@jerris.com>
	<CAEorZy=VFyVvfx+dCCfuBZvR6sS1tGgBfjnTZ_CUj=vj4NS0cA@mail.gmail.com>
	<CAKbxfG-_ouQdkOra2vkPgsoOpkPF6BmuHYzfDMN1BnQSmJfBSg@mail.gmail.com>
	<CAEorZykBsfBcS2-MMDdCeJak+ra3Oy2cSjm64Us+txGeyQhadQ@mail.gmail.com>
	<B319A4CC-3F70-4B58-93B2-556A291F3D27@jerris.com>
Message-ID: <CAEorZykBBssx9tsUi3VFqHg2Qh6U9kLBVnfVP2_Ljc=jJ4EDFQ@mail.gmail.com>

Other features of mod_av are working fine.  For example, I am to record to
an rtmp endpoint and see video.  Video codecs are negotiating properly.

On Thu, Jun 23, 2016 at 1:54 PM Michael Jerris <mike at jerris.com> wrote:

> I would confirm for sure thats installed, then look at the debug and
> confirm its actually negotiating a video codec.  Its really hard to make
> guesses with no information to go on so we are just making wild guesses
> here.
>
> On Jun 23, 2016, at 1:43 PM, Danny Gershman <danny.gershman at gmail.com>
> wrote:
>
> Already have it there, FS was compiled with it.
>
> On Thu, Jun 23, 2016 at 1:29 PM Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Like mike said, make sure you have main contrib non-free for you repos
>> and
>>
>> libavcodec-extra
>>
>>
>>
>> On Thu, Jun 23, 2016 at 12:12 PM, Danny Gershman <
>> danny.gershman at gmail.com> wrote:
>>
>>> I'm trying on 1.6.9 and I'm still not getting results.  Is there some
>>> specific condition in which this will work.  If I have muxing turned on,
>>> how does that display back to a softphone?  Should this be played in a
>>> specific channel?  Is playing a video file to all users in a conference,
>>> what would the expected response be?
>>>
>>> On Thu, Jun 16, 2016 at 8:08 PM Michael Jerris <mike at jerris.com> wrote:
>>>
>>>> Try 1.6.9 as thats now released, if thats not working, take a look at
>>>> the debug logs and see what it says.  One common issue is playing mp4 files
>>>> requires the libav-extras package on debian.
>>>>
>>>>
>>>> On Jun 16, 2016, at 6:54 PM, Danny Gershman <danny.gershman at gmail.com>
>>>> wrote:
>>>>
>>>> I'm having an issue playing back video in a conference, running (1.6.8
>>>> tag and loaded up mod_av), which it only plays audio back (no visual).  I'm
>>>> dialed in using a softphone with VP8 codec.
>>>>
>>>> I run command:
>>>>
>>>> conference <conf> play http://example.org/play.mp4
>>>>
>>>> Any pointers.
>>>>
>>>> Thanks,
>>>> Danny Gershman
>>>>
>>>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From krice at freeswitch.org  Thu Jun 23 21:57:56 2016
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 23 Jun 2016 12:57:56 -0500
Subject: [Freeswitch-users] Debian 8 package i386?
In-Reply-To: <04e201d1cd78$5b890110$129b0330$@secureachsystems.com>
References: <04e201d1cd78$5b890110$129b0330$@secureachsystems.com>
Message-ID: <77c301d1cd78$c66ce550$5346aff0$@freeswitch.org>

There is a known bug on the 32bit issues that will probably be corrected in
the next release.

 

Personally I don't recommend the 32bit x86 versions unless you are just
running on actually 32bit hardware.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason
Moran
Sent: Thursday, June 23, 2016 12:55 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Debian 8 package i386?

 

I have VMs of Deb8 for both 64bit and 32bit/i386. 
The instructions work great for 64bit, and FS installed as expected.

 

However, on i386 the command

# apt-get update && apt-get install -y freeswitch-meta-all

returns

E: Unable to locate package freeswitch-meta-all

 

I looked in
http://files.freeswitch.org/repo/deb/freeswitch-1.6/db/packages.db and see
i386 packages for 1.4 (Wheezy) but for 1.6 I only see amd64.

Is our old x86 server unsupported now?

 

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From danny.gershman at gmail.com  Thu Jun 23 21:59:19 2016
From: danny.gershman at gmail.com (Danny Gershman)
Date: Thu, 23 Jun 2016 17:59:19 +0000
Subject: [Freeswitch-users] conference play video
In-Reply-To: <CAEorZykBBssx9tsUi3VFqHg2Qh6U9kLBVnfVP2_Ljc=jJ4EDFQ@mail.gmail.com>
References: <CAEorZymOmq5b88qBgvCEO5-eoucLx99Si-cYu4HNKi4-Su16fg@mail.gmail.com>
	<618BE480-F3CD-4903-85A5-923FD12C2941@jerris.com>
	<CAEorZy=VFyVvfx+dCCfuBZvR6sS1tGgBfjnTZ_CUj=vj4NS0cA@mail.gmail.com>
	<CAKbxfG-_ouQdkOra2vkPgsoOpkPF6BmuHYzfDMN1BnQSmJfBSg@mail.gmail.com>
	<CAEorZykBsfBcS2-MMDdCeJak+ra3Oy2cSjm64Us+txGeyQhadQ@mail.gmail.com>
	<B319A4CC-3F70-4B58-93B2-556A291F3D27@jerris.com>
	<CAEorZykBBssx9tsUi3VFqHg2Qh6U9kLBVnfVP2_Ljc=jJ4EDFQ@mail.gmail.com>
Message-ID: <CAEorZynxUgGOqugfr3O=UN3EgGWjvojMckM=6kzzCFWKFTu8kw@mail.gmail.com>

Here is the output of the command I'm executing:

freeswitch at freeswitch>conference ~video2~conf_123 play
http://download.openbricks.org/sample/H264/h264_sintel_trailer-1080p.mp4
(play) Playing file
http://download.openbricks.org/sample/H264/h264_sintel_trailer-1080p.mp4
2016-06-23 04:37:23.278786 [DEBUG] avformat.c:1694 sample rate: 48000,
channels: 2
2016-06-23 04:37:23.450105 [INFO] avformat.c:1303 file
/usr/local/freeswitch/storage/http_file_cache/43772c747785c73f5beb03cc76d617de.mp4
is seekable
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from
'/usr/local/freeswitch/storage/http_file_cache/43772c747785c73f5beb03cc76d617de.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    creation_time   : 1970-01-01 00:00:00
    title           : Sintel Trailer
    artist          : Durian Open Movie Team
    encoder         : Lavf52.62.0
    copyright       : (c) copyright Blender Foundation | durian.blender.org
    description     : Trailer for the Sintel open movie project
  Duration:
00:00:52.20
, start:
0.000000
, bitrate:
2240 kb/s
    Stream #0.0
(und)
: Video: h264 (High), yuv420p, 1920x1080, 2108 kb/s
, 24 fps
, 24 tbn
, 48 tbc
 (default)
    Metadata:
      creation_time   : 1970-01-01 00:00:00
    Stream #0.1
(und)
: Audio: aac, 48000 Hz, stereo, fltp, 126 kb/s
 (default)
    Metadata:
freeswitch at freeswitch>
      creation_time   : 1970-01-01 00:00:00

On Thu, Jun 23, 2016 at 1:55 PM Danny Gershman <danny.gershman at gmail.com>
wrote:

> Other features of mod_av are working fine.  For example, I am to record to
> an rtmp endpoint and see video.  Video codecs are negotiating properly.
>
> On Thu, Jun 23, 2016 at 1:54 PM Michael Jerris <mike at jerris.com> wrote:
>
>> I would confirm for sure thats installed, then look at the debug and
>> confirm its actually negotiating a video codec.  Its really hard to make
>> guesses with no information to go on so we are just making wild guesses
>> here.
>>
>> On Jun 23, 2016, at 1:43 PM, Danny Gershman <danny.gershman at gmail.com>
>> wrote:
>>
>> Already have it there, FS was compiled with it.
>>
>> On Thu, Jun 23, 2016 at 1:29 PM Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> Like mike said, make sure you have main contrib non-free for you repos
>>> and
>>>
>>> libavcodec-extra
>>>
>>>
>>>
>>> On Thu, Jun 23, 2016 at 12:12 PM, Danny Gershman <
>>> danny.gershman at gmail.com> wrote:
>>>
>>>> I'm trying on 1.6.9 and I'm still not getting results.  Is there some
>>>> specific condition in which this will work.  If I have muxing turned on,
>>>> how does that display back to a softphone?  Should this be played in a
>>>> specific channel?  Is playing a video file to all users in a conference,
>>>> what would the expected response be?
>>>>
>>>> On Thu, Jun 16, 2016 at 8:08 PM Michael Jerris <mike at jerris.com> wrote:
>>>>
>>>>> Try 1.6.9 as thats now released, if thats not working, take a look at
>>>>> the debug logs and see what it says.  One common issue is playing mp4 files
>>>>> requires the libav-extras package on debian.
>>>>>
>>>>>
>>>>> On Jun 16, 2016, at 6:54 PM, Danny Gershman <danny.gershman at gmail.com>
>>>>> wrote:
>>>>>
>>>>> I'm having an issue playing back video in a conference, running (1.6.8
>>>>> tag and loaded up mod_av), which it only plays audio back (no visual).  I'm
>>>>> dialed in using a softphone with VP8 codec.
>>>>>
>>>>> I run command:
>>>>>
>>>>> conference <conf> play http://example.org/play.mp4
>>>>>
>>>>> Any pointers.
>>>>>
>>>>> Thanks,
>>>>> Danny Gershman
>>>>>
>>>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
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From mike at jerris.com  Thu Jun 23 22:00:51 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 23 Jun 2016 14:00:51 -0400
Subject: [Freeswitch-users] Debian 8 package i386?
In-Reply-To: <04e201d1cd78$5b890110$129b0330$@secureachsystems.com>
References: <04e201d1cd78$5b890110$129b0330$@secureachsystems.com>
Message-ID: <0ED1D0EF-3192-4113-8386-A3F9E4B52FD1@jerris.com>

we have an issue open with 32-bit packages due to a dependency issue related to sangoma code that isn?t packaged right.  As we have not a lot of people using 32bit it hasn?t been a very high priority to resolve, but its on the list to address.  The code builds fine on 32 but we won?t have new packages until we resolve that issue.


> On Jun 23, 2016, at 1:54 PM, Jason Moran <jmoran at secureachsystems.com> wrote:
> 
> I have VMs of Deb8 for both 64bit and 32bit/i386. 
> The instructions work great for 64bit, and FS installed as expected.
>  
> However, on i386 the command
> # apt-get update && apt-get install -y freeswitch-meta-all
> returns
> E: Unable to locate package freeswitch-meta-all
>  
> I looked in http://files.freeswitch.org/repo/deb/freeswitch-1.6/db/packages.db <http://files.freeswitch.org/repo/deb/freeswitch-1.6/db/packages.db> and see i386 packages for 1.4 (Wheezy) but for 1.6 I only see amd64.
> Is our old x86 server unsupported now?
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
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From danny.gershman at gmail.com  Thu Jun 23 23:01:43 2016
From: danny.gershman at gmail.com (Danny Gershman)
Date: Thu, 23 Jun 2016 19:01:43 +0000
Subject: [Freeswitch-users] conference play video
In-Reply-To: <CAEorZynxUgGOqugfr3O=UN3EgGWjvojMckM=6kzzCFWKFTu8kw@mail.gmail.com>
References: <CAEorZymOmq5b88qBgvCEO5-eoucLx99Si-cYu4HNKi4-Su16fg@mail.gmail.com>
	<618BE480-F3CD-4903-85A5-923FD12C2941@jerris.com>
	<CAEorZy=VFyVvfx+dCCfuBZvR6sS1tGgBfjnTZ_CUj=vj4NS0cA@mail.gmail.com>
	<CAKbxfG-_ouQdkOra2vkPgsoOpkPF6BmuHYzfDMN1BnQSmJfBSg@mail.gmail.com>
	<CAEorZykBsfBcS2-MMDdCeJak+ra3Oy2cSjm64Us+txGeyQhadQ@mail.gmail.com>
	<B319A4CC-3F70-4B58-93B2-556A291F3D27@jerris.com>
	<CAEorZykBBssx9tsUi3VFqHg2Qh6U9kLBVnfVP2_Ljc=jJ4EDFQ@mail.gmail.com>
	<CAEorZynxUgGOqugfr3O=UN3EgGWjvojMckM=6kzzCFWKFTu8kw@mail.gmail.com>
Message-ID: <CAEorZymaEWTTeH-4xnGiYF91poq_RXWzHNXUbeX9A+xXwkW+3A@mail.gmail.com>

I got this working on a video recorded by FS, but apparently this
particular video I was testing with didn't work too well.  Is there a
limitation on the videos that can be played back?  Some setting in the
conference.conf.xml?

On Thu, Jun 23, 2016 at 1:59 PM Danny Gershman <danny.gershman at gmail.com>
wrote:

> Here is the output of the command I'm executing:
>
> freeswitch at freeswitch>conference ~video2~conf_123 play
> http://download.openbricks.org/sample/H264/h264_sintel_trailer-1080p.mp4
> (play) Playing file
> http://download.openbricks.org/sample/H264/h264_sintel_trailer-1080p.mp4
> 2016-06-23 04:37:23.278786 [DEBUG] avformat.c:1694 sample rate: 48000,
> channels: 2
> 2016-06-23 04:37:23.450105 [INFO] avformat.c:1303 file
> /usr/local/freeswitch/storage/http_file_cache/43772c747785c73f5beb03cc76d617de.mp4
> is seekable
> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from
> '/usr/local/freeswitch/storage/http_file_cache/43772c747785c73f5beb03cc76d617de.mp4':
>   Metadata:
>     major_brand     : isom
>     minor_version   : 512
>     compatible_brands: isomiso2avc1mp41
>     creation_time   : 1970-01-01 00:00:00
>     title           : Sintel Trailer
>     artist          : Durian Open Movie Team
>     encoder         : Lavf52.62.0
>     copyright       : (c) copyright Blender Foundation |
> durian.blender.org
>     description     : Trailer for the Sintel open movie project
>   Duration:
> 00:00:52.20
> , start:
> 0.000000
> , bitrate:
> 2240 kb/s
>     Stream #0.0
> (und)
> : Video: h264 (High), yuv420p, 1920x1080, 2108 kb/s
> , 24 fps
> , 24 tbn
> , 48 tbc
>  (default)
>     Metadata:
>       creation_time   : 1970-01-01 00:00:00
>     Stream #0.1
> (und)
> : Audio: aac, 48000 Hz, stereo, fltp, 126 kb/s
>  (default)
>     Metadata:
> freeswitch at freeswitch>
>       creation_time   : 1970-01-01 00:00:00
>
> On Thu, Jun 23, 2016 at 1:55 PM Danny Gershman <danny.gershman at gmail.com>
> wrote:
>
>> Other features of mod_av are working fine.  For example, I am to record
>> to an rtmp endpoint and see video.  Video codecs are negotiating properly.
>>
>> On Thu, Jun 23, 2016 at 1:54 PM Michael Jerris <mike at jerris.com> wrote:
>>
>>> I would confirm for sure thats installed, then look at the debug and
>>> confirm its actually negotiating a video codec.  Its really hard to make
>>> guesses with no information to go on so we are just making wild guesses
>>> here.
>>>
>>> On Jun 23, 2016, at 1:43 PM, Danny Gershman <danny.gershman at gmail.com>
>>> wrote:
>>>
>>> Already have it there, FS was compiled with it.
>>>
>>> On Thu, Jun 23, 2016 at 1:29 PM Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> Like mike said, make sure you have main contrib non-free for you repos
>>>> and
>>>>
>>>> libavcodec-extra
>>>>
>>>>
>>>>
>>>> On Thu, Jun 23, 2016 at 12:12 PM, Danny Gershman <
>>>> danny.gershman at gmail.com> wrote:
>>>>
>>>>> I'm trying on 1.6.9 and I'm still not getting results.  Is there some
>>>>> specific condition in which this will work.  If I have muxing turned on,
>>>>> how does that display back to a softphone?  Should this be played in a
>>>>> specific channel?  Is playing a video file to all users in a conference,
>>>>> what would the expected response be?
>>>>>
>>>>> On Thu, Jun 16, 2016 at 8:08 PM Michael Jerris <mike at jerris.com>
>>>>> wrote:
>>>>>
>>>>>> Try 1.6.9 as thats now released, if thats not working, take a look at
>>>>>> the debug logs and see what it says.  One common issue is playing mp4 files
>>>>>> requires the libav-extras package on debian.
>>>>>>
>>>>>>
>>>>>> On Jun 16, 2016, at 6:54 PM, Danny Gershman <danny.gershman at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>> I'm having an issue playing back video in a conference, running
>>>>>> (1.6.8 tag and loaded up mod_av), which it only plays audio back (no
>>>>>> visual).  I'm dialed in using a softphone with VP8 codec.
>>>>>>
>>>>>> I run command:
>>>>>>
>>>>>> conference <conf> play http://example.org/play.mp4
>>>>>>
>>>>>> Any pointers.
>>>>>>
>>>>>> Thanks,
>>>>>> Danny Gershman
>>>>>>
>>>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
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From anthony.minessale at gmail.com  Fri Jun 24 00:06:10 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 23 Jun 2016 15:06:10 -0500
Subject: [Freeswitch-users] conference play video
In-Reply-To: <CAEorZymaEWTTeH-4xnGiYF91poq_RXWzHNXUbeX9A+xXwkW+3A@mail.gmail.com>
References: <CAEorZymOmq5b88qBgvCEO5-eoucLx99Si-cYu4HNKi4-Su16fg@mail.gmail.com>
	<618BE480-F3CD-4903-85A5-923FD12C2941@jerris.com>
	<CAEorZy=VFyVvfx+dCCfuBZvR6sS1tGgBfjnTZ_CUj=vj4NS0cA@mail.gmail.com>
	<CAKbxfG-_ouQdkOra2vkPgsoOpkPF6BmuHYzfDMN1BnQSmJfBSg@mail.gmail.com>
	<CAEorZykBsfBcS2-MMDdCeJak+ra3Oy2cSjm64Us+txGeyQhadQ@mail.gmail.com>
	<B319A4CC-3F70-4B58-93B2-556A291F3D27@jerris.com>
	<CAEorZykBBssx9tsUi3VFqHg2Qh6U9kLBVnfVP2_Ljc=jJ4EDFQ@mail.gmail.com>
	<CAEorZynxUgGOqugfr3O=UN3EgGWjvojMckM=6kzzCFWKFTu8kw@mail.gmail.com>
	<CAEorZymaEWTTeH-4xnGiYF91poq_RXWzHNXUbeX9A+xXwkW+3A@mail.gmail.com>
Message-ID: <CAKbxfG89-1kPhQh-Y6YAx8GYh7_udTsZxyqbuYSsCaadmprqHQ@mail.gmail.com>

Those details are relevant.

I tried it and i see the same problem but only when using the http url.
If I wget the file then it place so we may have a small bug with playing
mp4 using htttapi.  I have never tried this before.


On Thu, Jun 23, 2016 at 2:01 PM, Danny Gershman <danny.gershman at gmail.com>
wrote:

> I got this working on a video recorded by FS, but apparently this
> particular video I was testing with didn't work too well.  Is there a
> limitation on the videos that can be played back?  Some setting in the
> conference.conf.xml?
>
> On Thu, Jun 23, 2016 at 1:59 PM Danny Gershman <danny.gershman at gmail.com>
> wrote:
>
>> Here is the output of the command I'm executing:
>>
>> freeswitch at freeswitch>conference ~video2~conf_123 play
>> http://download.openbricks.org/sample/H264/h264_sintel_trailer-1080p.mp4
>> (play) Playing file
>> http://download.openbricks.org/sample/H264/h264_sintel_trailer-1080p.mp4
>> 2016-06-23 04:37:23.278786 [DEBUG] avformat.c:1694 sample rate: 48000,
>> channels: 2
>> 2016-06-23 04:37:23.450105 [INFO] avformat.c:1303 file
>> /usr/local/freeswitch/storage/http_file_cache/43772c747785c73f5beb03cc76d617de.mp4
>> is seekable
>> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from
>> '/usr/local/freeswitch/storage/http_file_cache/43772c747785c73f5beb03cc76d617de.mp4':
>>   Metadata:
>>     major_brand     : isom
>>     minor_version   : 512
>>     compatible_brands: isomiso2avc1mp41
>>     creation_time   : 1970-01-01 00:00:00
>>     title           : Sintel Trailer
>>     artist          : Durian Open Movie Team
>>     encoder         : Lavf52.62.0
>>     copyright       : (c) copyright Blender Foundation |
>> durian.blender.org
>>     description     : Trailer for the Sintel open movie project
>>   Duration:
>> 00:00:52.20
>> , start:
>> 0.000000
>> , bitrate:
>> 2240 kb/s
>>     Stream #0.0
>> (und)
>> : Video: h264 (High), yuv420p, 1920x1080, 2108 kb/s
>> , 24 fps
>> , 24 tbn
>> , 48 tbc
>>  (default)
>>     Metadata:
>>       creation_time   : 1970-01-01 00:00:00
>>     Stream #0.1
>> (und)
>> : Audio: aac, 48000 Hz, stereo, fltp, 126 kb/s
>>  (default)
>>     Metadata:
>> freeswitch at freeswitch>
>>       creation_time   : 1970-01-01 00:00:00
>>
>> On Thu, Jun 23, 2016 at 1:55 PM Danny Gershman <danny.gershman at gmail.com>
>> wrote:
>>
>>> Other features of mod_av are working fine.  For example, I am to record
>>> to an rtmp endpoint and see video.  Video codecs are negotiating properly.
>>>
>>> On Thu, Jun 23, 2016 at 1:54 PM Michael Jerris <mike at jerris.com> wrote:
>>>
>>>> I would confirm for sure thats installed, then look at the debug and
>>>> confirm its actually negotiating a video codec.  Its really hard to make
>>>> guesses with no information to go on so we are just making wild guesses
>>>> here.
>>>>
>>>> On Jun 23, 2016, at 1:43 PM, Danny Gershman <danny.gershman at gmail.com>
>>>> wrote:
>>>>
>>>> Already have it there, FS was compiled with it.
>>>>
>>>> On Thu, Jun 23, 2016 at 1:29 PM Anthony Minessale <
>>>> anthony.minessale at gmail.com> wrote:
>>>>
>>>>> Like mike said, make sure you have main contrib non-free for you repos
>>>>> and
>>>>>
>>>>> libavcodec-extra
>>>>>
>>>>>
>>>>>
>>>>> On Thu, Jun 23, 2016 at 12:12 PM, Danny Gershman <
>>>>> danny.gershman at gmail.com> wrote:
>>>>>
>>>>>> I'm trying on 1.6.9 and I'm still not getting results.  Is there some
>>>>>> specific condition in which this will work.  If I have muxing turned on,
>>>>>> how does that display back to a softphone?  Should this be played in a
>>>>>> specific channel?  Is playing a video file to all users in a conference,
>>>>>> what would the expected response be?
>>>>>>
>>>>>> On Thu, Jun 16, 2016 at 8:08 PM Michael Jerris <mike at jerris.com>
>>>>>> wrote:
>>>>>>
>>>>>>> Try 1.6.9 as thats now released, if thats not working, take a look
>>>>>>> at the debug logs and see what it says.  One common issue is playing mp4
>>>>>>> files requires the libav-extras package on debian.
>>>>>>>
>>>>>>>
>>>>>>> On Jun 16, 2016, at 6:54 PM, Danny Gershman <
>>>>>>> danny.gershman at gmail.com> wrote:
>>>>>>>
>>>>>>> I'm having an issue playing back video in a conference, running
>>>>>>> (1.6.8 tag and loaded up mod_av), which it only plays audio back (no
>>>>>>> visual).  I'm dialed in using a softphone with VP8 codec.
>>>>>>>
>>>>>>> I run command:
>>>>>>>
>>>>>>> conference <conf> play http://example.org/play.mp4
>>>>>>>
>>>>>>> Any pointers.
>>>>>>>
>>>>>>> Thanks,
>>>>>>> Danny Gershman
>>>>>>>
>>>>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From danny.gershman at gmail.com  Fri Jun 24 00:12:23 2016
From: danny.gershman at gmail.com (Danny Gershman)
Date: Thu, 23 Jun 2016 20:12:23 +0000
Subject: [Freeswitch-users] conference play video
In-Reply-To: <CAKbxfG89-1kPhQh-Y6YAx8GYh7_udTsZxyqbuYSsCaadmprqHQ@mail.gmail.com>
References: <CAEorZymOmq5b88qBgvCEO5-eoucLx99Si-cYu4HNKi4-Su16fg@mail.gmail.com>
	<618BE480-F3CD-4903-85A5-923FD12C2941@jerris.com>
	<CAEorZy=VFyVvfx+dCCfuBZvR6sS1tGgBfjnTZ_CUj=vj4NS0cA@mail.gmail.com>
	<CAKbxfG-_ouQdkOra2vkPgsoOpkPF6BmuHYzfDMN1BnQSmJfBSg@mail.gmail.com>
	<CAEorZykBsfBcS2-MMDdCeJak+ra3Oy2cSjm64Us+txGeyQhadQ@mail.gmail.com>
	<B319A4CC-3F70-4B58-93B2-556A291F3D27@jerris.com>
	<CAEorZykBBssx9tsUi3VFqHg2Qh6U9kLBVnfVP2_Ljc=jJ4EDFQ@mail.gmail.com>
	<CAEorZynxUgGOqugfr3O=UN3EgGWjvojMckM=6kzzCFWKFTu8kw@mail.gmail.com>
	<CAEorZymaEWTTeH-4xnGiYF91poq_RXWzHNXUbeX9A+xXwkW+3A@mail.gmail.com>
	<CAKbxfG89-1kPhQh-Y6YAx8GYh7_udTsZxyqbuYSsCaadmprqHQ@mail.gmail.com>
Message-ID: <CAEorZykV2hMkmQdEJ5-wDjZ3yg1H8rKyKPnWkeCyfwjbk2_f6g@mail.gmail.com>

I am able to see the same on my side, video file is playing now.  Thanks.

On Thu, Jun 23, 2016 at 4:07 PM Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Those details are relevant.
>
> I tried it and i see the same problem but only when using the http url.
> If I wget the file then it place so we may have a small bug with playing
> mp4 using htttapi.  I have never tried this before.
>
>
> On Thu, Jun 23, 2016 at 2:01 PM, Danny Gershman <danny.gershman at gmail.com>
> wrote:
>
>> I got this working on a video recorded by FS, but apparently this
>> particular video I was testing with didn't work too well.  Is there a
>> limitation on the videos that can be played back?  Some setting in the
>> conference.conf.xml?
>>
>> On Thu, Jun 23, 2016 at 1:59 PM Danny Gershman <danny.gershman at gmail.com>
>> wrote:
>>
>>> Here is the output of the command I'm executing:
>>>
>>> freeswitch at freeswitch>conference ~video2~conf_123 play
>>> http://download.openbricks.org/sample/H264/h264_sintel_trailer-1080p.mp4
>>> (play) Playing file
>>> http://download.openbricks.org/sample/H264/h264_sintel_trailer-1080p.mp4
>>> 2016-06-23 04:37:23.278786 [DEBUG] avformat.c:1694 sample rate: 48000,
>>> channels: 2
>>> 2016-06-23 04:37:23.450105 [INFO] avformat.c:1303 file
>>> /usr/local/freeswitch/storage/http_file_cache/43772c747785c73f5beb03cc76d617de.mp4
>>> is seekable
>>> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from
>>> '/usr/local/freeswitch/storage/http_file_cache/43772c747785c73f5beb03cc76d617de.mp4':
>>>   Metadata:
>>>     major_brand     : isom
>>>     minor_version   : 512
>>>     compatible_brands: isomiso2avc1mp41
>>>     creation_time   : 1970-01-01 00:00:00
>>>     title           : Sintel Trailer
>>>     artist          : Durian Open Movie Team
>>>     encoder         : Lavf52.62.0
>>>     copyright       : (c) copyright Blender Foundation |
>>> durian.blender.org
>>>     description     : Trailer for the Sintel open movie project
>>>   Duration:
>>> 00:00:52.20
>>> , start:
>>> 0.000000
>>> , bitrate:
>>> 2240 kb/s
>>>     Stream #0.0
>>> (und)
>>> : Video: h264 (High), yuv420p, 1920x1080, 2108 kb/s
>>> , 24 fps
>>> , 24 tbn
>>> , 48 tbc
>>>  (default)
>>>     Metadata:
>>>       creation_time   : 1970-01-01 00:00:00
>>>     Stream #0.1
>>> (und)
>>> : Audio: aac, 48000 Hz, stereo, fltp, 126 kb/s
>>>  (default)
>>>     Metadata:
>>> freeswitch at freeswitch>
>>>       creation_time   : 1970-01-01 00:00:00
>>>
>>> On Thu, Jun 23, 2016 at 1:55 PM Danny Gershman <danny.gershman at gmail.com>
>>> wrote:
>>>
>>>> Other features of mod_av are working fine.  For example, I am to record
>>>> to an rtmp endpoint and see video.  Video codecs are negotiating properly.
>>>>
>>>> On Thu, Jun 23, 2016 at 1:54 PM Michael Jerris <mike at jerris.com> wrote:
>>>>
>>>>> I would confirm for sure thats installed, then look at the debug and
>>>>> confirm its actually negotiating a video codec.  Its really hard to make
>>>>> guesses with no information to go on so we are just making wild guesses
>>>>> here.
>>>>>
>>>>> On Jun 23, 2016, at 1:43 PM, Danny Gershman <danny.gershman at gmail.com>
>>>>> wrote:
>>>>>
>>>>> Already have it there, FS was compiled with it.
>>>>>
>>>>> On Thu, Jun 23, 2016 at 1:29 PM Anthony Minessale <
>>>>> anthony.minessale at gmail.com> wrote:
>>>>>
>>>>>> Like mike said, make sure you have main contrib non-free for you
>>>>>> repos and
>>>>>>
>>>>>> libavcodec-extra
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Thu, Jun 23, 2016 at 12:12 PM, Danny Gershman <
>>>>>> danny.gershman at gmail.com> wrote:
>>>>>>
>>>>>>> I'm trying on 1.6.9 and I'm still not getting results.  Is there
>>>>>>> some specific condition in which this will work.  If I have muxing turned
>>>>>>> on, how does that display back to a softphone?  Should this be played in a
>>>>>>> specific channel?  Is playing a video file to all users in a conference,
>>>>>>> what would the expected response be?
>>>>>>>
>>>>>>> On Thu, Jun 16, 2016 at 8:08 PM Michael Jerris <mike at jerris.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> Try 1.6.9 as thats now released, if thats not working, take a look
>>>>>>>> at the debug logs and see what it says.  One common issue is playing mp4
>>>>>>>> files requires the libav-extras package on debian.
>>>>>>>>
>>>>>>>>
>>>>>>>> On Jun 16, 2016, at 6:54 PM, Danny Gershman <
>>>>>>>> danny.gershman at gmail.com> wrote:
>>>>>>>>
>>>>>>>> I'm having an issue playing back video in a conference, running
>>>>>>>> (1.6.8 tag and loaded up mod_av), which it only plays audio back (no
>>>>>>>> visual).  I'm dialed in using a softphone with VP8 codec.
>>>>>>>>
>>>>>>>> I run command:
>>>>>>>>
>>>>>>>> conference <conf> play http://example.org/play.mp4
>>>>>>>>
>>>>>>>> Any pointers.
>>>>>>>>
>>>>>>>> Thanks,
>>>>>>>> Danny Gershman
>>>>>>>>
>>>>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From anthony.minessale at gmail.com  Fri Jun 24 01:04:54 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 23 Jun 2016 16:04:54 -0500
Subject: [Freeswitch-users] conference play video
In-Reply-To: <CAEorZykV2hMkmQdEJ5-wDjZ3yg1H8rKyKPnWkeCyfwjbk2_f6g@mail.gmail.com>
References: <CAEorZymOmq5b88qBgvCEO5-eoucLx99Si-cYu4HNKi4-Su16fg@mail.gmail.com>
	<618BE480-F3CD-4903-85A5-923FD12C2941@jerris.com>
	<CAEorZy=VFyVvfx+dCCfuBZvR6sS1tGgBfjnTZ_CUj=vj4NS0cA@mail.gmail.com>
	<CAKbxfG-_ouQdkOra2vkPgsoOpkPF6BmuHYzfDMN1BnQSmJfBSg@mail.gmail.com>
	<CAEorZykBsfBcS2-MMDdCeJak+ra3Oy2cSjm64Us+txGeyQhadQ@mail.gmail.com>
	<B319A4CC-3F70-4B58-93B2-556A291F3D27@jerris.com>
	<CAEorZykBBssx9tsUi3VFqHg2Qh6U9kLBVnfVP2_Ljc=jJ4EDFQ@mail.gmail.com>
	<CAEorZynxUgGOqugfr3O=UN3EgGWjvojMckM=6kzzCFWKFTu8kw@mail.gmail.com>
	<CAEorZymaEWTTeH-4xnGiYF91poq_RXWzHNXUbeX9A+xXwkW+3A@mail.gmail.com>
	<CAKbxfG89-1kPhQh-Y6YAx8GYh7_udTsZxyqbuYSsCaadmprqHQ@mail.gmail.com>
	<CAEorZykV2hMkmQdEJ5-wDjZ3yg1H8rKyKPnWkeCyfwjbk2_f6g@mail.gmail.com>
Message-ID: <CAKbxfG-Yi+YqbSgJ9_9AF_BLxncivUyeY4ggVygidtSHMrnmDQ@mail.gmail.com>

https://freeswitch.org/jira/browse/FS-9296

Update to master and it should work now.


On Thu, Jun 23, 2016 at 3:12 PM, Danny Gershman <danny.gershman at gmail.com>
wrote:

> I am able to see the same on my side, video file is playing now.  Thanks.
>
> On Thu, Jun 23, 2016 at 4:07 PM Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Those details are relevant.
>>
>> I tried it and i see the same problem but only when using the http url.
>> If I wget the file then it place so we may have a small bug with playing
>> mp4 using htttapi.  I have never tried this before.
>>
>>
>> On Thu, Jun 23, 2016 at 2:01 PM, Danny Gershman <danny.gershman at gmail.com
>> > wrote:
>>
>>> I got this working on a video recorded by FS, but apparently this
>>> particular video I was testing with didn't work too well.  Is there a
>>> limitation on the videos that can be played back?  Some setting in the
>>> conference.conf.xml?
>>>
>>> On Thu, Jun 23, 2016 at 1:59 PM Danny Gershman <danny.gershman at gmail.com>
>>> wrote:
>>>
>>>> Here is the output of the command I'm executing:
>>>>
>>>> freeswitch at freeswitch>conference ~video2~conf_123 play
>>>> http://download.openbricks.org/sample/H264/h264_sintel_trailer-1080p.mp4
>>>> (play) Playing file
>>>> http://download.openbricks.org/sample/H264/h264_sintel_trailer-1080p.mp4
>>>> 2016-06-23 04:37:23.278786 [DEBUG] avformat.c:1694 sample rate: 48000,
>>>> channels: 2
>>>> 2016-06-23 04:37:23.450105 [INFO] avformat.c:1303 file
>>>> /usr/local/freeswitch/storage/http_file_cache/43772c747785c73f5beb03cc76d617de.mp4
>>>> is seekable
>>>> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from
>>>> '/usr/local/freeswitch/storage/http_file_cache/43772c747785c73f5beb03cc76d617de.mp4':
>>>>   Metadata:
>>>>     major_brand     : isom
>>>>     minor_version   : 512
>>>>     compatible_brands: isomiso2avc1mp41
>>>>     creation_time   : 1970-01-01 00:00:00
>>>>     title           : Sintel Trailer
>>>>     artist          : Durian Open Movie Team
>>>>     encoder         : Lavf52.62.0
>>>>     copyright       : (c) copyright Blender Foundation |
>>>> durian.blender.org
>>>>     description     : Trailer for the Sintel open movie project
>>>>   Duration:
>>>> 00:00:52.20
>>>> , start:
>>>> 0.000000
>>>> , bitrate:
>>>> 2240 kb/s
>>>>     Stream #0.0
>>>> (und)
>>>> : Video: h264 (High), yuv420p, 1920x1080, 2108 kb/s
>>>> , 24 fps
>>>> , 24 tbn
>>>> , 48 tbc
>>>>  (default)
>>>>     Metadata:
>>>>       creation_time   : 1970-01-01 00:00:00
>>>>     Stream #0.1
>>>> (und)
>>>> : Audio: aac, 48000 Hz, stereo, fltp, 126 kb/s
>>>>  (default)
>>>>     Metadata:
>>>> freeswitch at freeswitch>
>>>>       creation_time   : 1970-01-01 00:00:00
>>>>
>>>> On Thu, Jun 23, 2016 at 1:55 PM Danny Gershman <
>>>> danny.gershman at gmail.com> wrote:
>>>>
>>>>> Other features of mod_av are working fine.  For example, I am to
>>>>> record to an rtmp endpoint and see video.  Video codecs are negotiating
>>>>> properly.
>>>>>
>>>>> On Thu, Jun 23, 2016 at 1:54 PM Michael Jerris <mike at jerris.com>
>>>>> wrote:
>>>>>
>>>>>> I would confirm for sure thats installed, then look at the debug and
>>>>>> confirm its actually negotiating a video codec.  Its really hard to make
>>>>>> guesses with no information to go on so we are just making wild guesses
>>>>>> here.
>>>>>>
>>>>>> On Jun 23, 2016, at 1:43 PM, Danny Gershman <danny.gershman at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>> Already have it there, FS was compiled with it.
>>>>>>
>>>>>> On Thu, Jun 23, 2016 at 1:29 PM Anthony Minessale <
>>>>>> anthony.minessale at gmail.com> wrote:
>>>>>>
>>>>>>> Like mike said, make sure you have main contrib non-free for you
>>>>>>> repos and
>>>>>>>
>>>>>>> libavcodec-extra
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Thu, Jun 23, 2016 at 12:12 PM, Danny Gershman <
>>>>>>> danny.gershman at gmail.com> wrote:
>>>>>>>
>>>>>>>> I'm trying on 1.6.9 and I'm still not getting results.  Is there
>>>>>>>> some specific condition in which this will work.  If I have muxing turned
>>>>>>>> on, how does that display back to a softphone?  Should this be played in a
>>>>>>>> specific channel?  Is playing a video file to all users in a conference,
>>>>>>>> what would the expected response be?
>>>>>>>>
>>>>>>>> On Thu, Jun 16, 2016 at 8:08 PM Michael Jerris <mike at jerris.com>
>>>>>>>> wrote:
>>>>>>>>
>>>>>>>>> Try 1.6.9 as thats now released, if thats not working, take a look
>>>>>>>>> at the debug logs and see what it says.  One common issue is playing mp4
>>>>>>>>> files requires the libav-extras package on debian.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Jun 16, 2016, at 6:54 PM, Danny Gershman <
>>>>>>>>> danny.gershman at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>> I'm having an issue playing back video in a conference, running
>>>>>>>>> (1.6.8 tag and loaded up mod_av), which it only plays audio back (no
>>>>>>>>> visual).  I'm dialed in using a softphone with VP8 codec.
>>>>>>>>>
>>>>>>>>> I run command:
>>>>>>>>>
>>>>>>>>> conference <conf> play http://example.org/play.mp4
>>>>>>>>>
>>>>>>>>> Any pointers.
>>>>>>>>>
>>>>>>>>> Thanks,
>>>>>>>>> Danny Gershman
>>>>>>>>>
>>>>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>>
>> ? http://freeswitch.org/  ? http://cluecon.com/  ?
>> http://twitter.com/FreeSWITCH
>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
>> <http://freeswitch.org/g+>*
>>
>> ClueCon Weekly Development Call
>> ? sip:888 at conference.freeswitch.org  ? +19193869900
>>
>> https://www.youtube.com/watch?v=9XXgW34t40s
>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
-------------- next part --------------
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From danny.gershman at gmail.com  Fri Jun 24 01:08:45 2016
From: danny.gershman at gmail.com (Danny Gershman)
Date: Thu, 23 Jun 2016 21:08:45 +0000
Subject: [Freeswitch-users] conference play video
In-Reply-To: <CAKbxfG-Yi+YqbSgJ9_9AF_BLxncivUyeY4ggVygidtSHMrnmDQ@mail.gmail.com>
References: <CAEorZymOmq5b88qBgvCEO5-eoucLx99Si-cYu4HNKi4-Su16fg@mail.gmail.com>
	<618BE480-F3CD-4903-85A5-923FD12C2941@jerris.com>
	<CAEorZy=VFyVvfx+dCCfuBZvR6sS1tGgBfjnTZ_CUj=vj4NS0cA@mail.gmail.com>
	<CAKbxfG-_ouQdkOra2vkPgsoOpkPF6BmuHYzfDMN1BnQSmJfBSg@mail.gmail.com>
	<CAEorZykBsfBcS2-MMDdCeJak+ra3Oy2cSjm64Us+txGeyQhadQ@mail.gmail.com>
	<B319A4CC-3F70-4B58-93B2-556A291F3D27@jerris.com>
	<CAEorZykBBssx9tsUi3VFqHg2Qh6U9kLBVnfVP2_Ljc=jJ4EDFQ@mail.gmail.com>
	<CAEorZynxUgGOqugfr3O=UN3EgGWjvojMckM=6kzzCFWKFTu8kw@mail.gmail.com>
	<CAEorZymaEWTTeH-4xnGiYF91poq_RXWzHNXUbeX9A+xXwkW+3A@mail.gmail.com>
	<CAKbxfG89-1kPhQh-Y6YAx8GYh7_udTsZxyqbuYSsCaadmprqHQ@mail.gmail.com>
	<CAEorZykV2hMkmQdEJ5-wDjZ3yg1H8rKyKPnWkeCyfwjbk2_f6g@mail.gmail.com>
	<CAKbxfG-Yi+YqbSgJ9_9AF_BLxncivUyeY4ggVygidtSHMrnmDQ@mail.gmail.com>
Message-ID: <CAEorZyn3HAXHka+_afUyDEuPg1xaSN=VC9xs89hyE=+UMGPVWw@mail.gmail.com>

I also noticed some weirdness when combining this with mod_http_cache.

On Thu, Jun 23, 2016 at 5:05 PM Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> https://freeswitch.org/jira/browse/FS-9296
>
> Update to master and it should work now.
>
>
> On Thu, Jun 23, 2016 at 3:12 PM, Danny Gershman <danny.gershman at gmail.com>
> wrote:
>
>> I am able to see the same on my side, video file is playing now.  Thanks.
>>
>> On Thu, Jun 23, 2016 at 4:07 PM Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> Those details are relevant.
>>>
>>> I tried it and i see the same problem but only when using the http url.
>>> If I wget the file then it place so we may have a small bug with playing
>>> mp4 using htttapi.  I have never tried this before.
>>>
>>>
>>> On Thu, Jun 23, 2016 at 2:01 PM, Danny Gershman <
>>> danny.gershman at gmail.com> wrote:
>>>
>>>> I got this working on a video recorded by FS, but apparently this
>>>> particular video I was testing with didn't work too well.  Is there a
>>>> limitation on the videos that can be played back?  Some setting in the
>>>> conference.conf.xml?
>>>>
>>>> On Thu, Jun 23, 2016 at 1:59 PM Danny Gershman <
>>>> danny.gershman at gmail.com> wrote:
>>>>
>>>>> Here is the output of the command I'm executing:
>>>>>
>>>>> freeswitch at freeswitch>conference ~video2~conf_123 play
>>>>> http://download.openbricks.org/sample/H264/h264_sintel_trailer-1080p.mp4
>>>>> (play) Playing file
>>>>> http://download.openbricks.org/sample/H264/h264_sintel_trailer-1080p.mp4
>>>>> 2016-06-23 04:37:23.278786 [DEBUG] avformat.c:1694 sample rate: 48000,
>>>>> channels: 2
>>>>> 2016-06-23 04:37:23.450105 [INFO] avformat.c:1303 file
>>>>> /usr/local/freeswitch/storage/http_file_cache/43772c747785c73f5beb03cc76d617de.mp4
>>>>> is seekable
>>>>> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from
>>>>> '/usr/local/freeswitch/storage/http_file_cache/43772c747785c73f5beb03cc76d617de.mp4':
>>>>>   Metadata:
>>>>>     major_brand     : isom
>>>>>     minor_version   : 512
>>>>>     compatible_brands: isomiso2avc1mp41
>>>>>     creation_time   : 1970-01-01 00:00:00
>>>>>     title           : Sintel Trailer
>>>>>     artist          : Durian Open Movie Team
>>>>>     encoder         : Lavf52.62.0
>>>>>     copyright       : (c) copyright Blender Foundation |
>>>>> durian.blender.org
>>>>>     description     : Trailer for the Sintel open movie project
>>>>>   Duration:
>>>>> 00:00:52.20
>>>>> , start:
>>>>> 0.000000
>>>>> , bitrate:
>>>>> 2240 kb/s
>>>>>     Stream #0.0
>>>>> (und)
>>>>> : Video: h264 (High), yuv420p, 1920x1080, 2108 kb/s
>>>>> , 24 fps
>>>>> , 24 tbn
>>>>> , 48 tbc
>>>>>  (default)
>>>>>     Metadata:
>>>>>       creation_time   : 1970-01-01 00:00:00
>>>>>     Stream #0.1
>>>>> (und)
>>>>> : Audio: aac, 48000 Hz, stereo, fltp, 126 kb/s
>>>>>  (default)
>>>>>     Metadata:
>>>>> freeswitch at freeswitch>
>>>>>       creation_time   : 1970-01-01 00:00:00
>>>>>
>>>>> On Thu, Jun 23, 2016 at 1:55 PM Danny Gershman <
>>>>> danny.gershman at gmail.com> wrote:
>>>>>
>>>>>> Other features of mod_av are working fine.  For example, I am to
>>>>>> record to an rtmp endpoint and see video.  Video codecs are negotiating
>>>>>> properly.
>>>>>>
>>>>>> On Thu, Jun 23, 2016 at 1:54 PM Michael Jerris <mike at jerris.com>
>>>>>> wrote:
>>>>>>
>>>>>>> I would confirm for sure thats installed, then look at the debug and
>>>>>>> confirm its actually negotiating a video codec.  Its really hard to make
>>>>>>> guesses with no information to go on so we are just making wild guesses
>>>>>>> here.
>>>>>>>
>>>>>>> On Jun 23, 2016, at 1:43 PM, Danny Gershman <
>>>>>>> danny.gershman at gmail.com> wrote:
>>>>>>>
>>>>>>> Already have it there, FS was compiled with it.
>>>>>>>
>>>>>>> On Thu, Jun 23, 2016 at 1:29 PM Anthony Minessale <
>>>>>>> anthony.minessale at gmail.com> wrote:
>>>>>>>
>>>>>>>> Like mike said, make sure you have main contrib non-free for you
>>>>>>>> repos and
>>>>>>>>
>>>>>>>> libavcodec-extra
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Thu, Jun 23, 2016 at 12:12 PM, Danny Gershman <
>>>>>>>> danny.gershman at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> I'm trying on 1.6.9 and I'm still not getting results.  Is there
>>>>>>>>> some specific condition in which this will work.  If I have muxing turned
>>>>>>>>> on, how does that display back to a softphone?  Should this be played in a
>>>>>>>>> specific channel?  Is playing a video file to all users in a conference,
>>>>>>>>> what would the expected response be?
>>>>>>>>>
>>>>>>>>> On Thu, Jun 16, 2016 at 8:08 PM Michael Jerris <mike at jerris.com>
>>>>>>>>> wrote:
>>>>>>>>>
>>>>>>>>>> Try 1.6.9 as thats now released, if thats not working, take a
>>>>>>>>>> look at the debug logs and see what it says.  One common issue is playing
>>>>>>>>>> mp4 files requires the libav-extras package on debian.
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Jun 16, 2016, at 6:54 PM, Danny Gershman <
>>>>>>>>>> danny.gershman at gmail.com> wrote:
>>>>>>>>>>
>>>>>>>>>> I'm having an issue playing back video in a conference, running
>>>>>>>>>> (1.6.8 tag and loaded up mod_av), which it only plays audio back (no
>>>>>>>>>> visual).  I'm dialed in using a softphone with VP8 codec.
>>>>>>>>>>
>>>>>>>>>> I run command:
>>>>>>>>>>
>>>>>>>>>> conference <conf> play http://example.org/play.mp4
>>>>>>>>>>
>>>>>>>>>> Any pointers.
>>>>>>>>>>
>>>>>>>>>> Thanks,
>>>>>>>>>> Danny Gershman
>>>>>>>>>>
>>>>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>>>
>>> ? http://freeswitch.org/  ? http://cluecon.com/  ?
>>> http://twitter.com/FreeSWITCH
>>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
>>> <http://freeswitch.org/g+>*
>>>
>>> ClueCon Weekly Development Call
>>> ? sip:888 at conference.freeswitch.org  ? +19193869900
>>>
>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From danny.gershman at gmail.com  Fri Jun 24 01:09:58 2016
From: danny.gershman at gmail.com (Danny Gershman)
Date: Thu, 23 Jun 2016 21:09:58 +0000
Subject: [Freeswitch-users] conference play video
In-Reply-To: <CAEorZyn3HAXHka+_afUyDEuPg1xaSN=VC9xs89hyE=+UMGPVWw@mail.gmail.com>
References: <CAEorZymOmq5b88qBgvCEO5-eoucLx99Si-cYu4HNKi4-Su16fg@mail.gmail.com>
	<618BE480-F3CD-4903-85A5-923FD12C2941@jerris.com>
	<CAEorZy=VFyVvfx+dCCfuBZvR6sS1tGgBfjnTZ_CUj=vj4NS0cA@mail.gmail.com>
	<CAKbxfG-_ouQdkOra2vkPgsoOpkPF6BmuHYzfDMN1BnQSmJfBSg@mail.gmail.com>
	<CAEorZykBsfBcS2-MMDdCeJak+ra3Oy2cSjm64Us+txGeyQhadQ@mail.gmail.com>
	<B319A4CC-3F70-4B58-93B2-556A291F3D27@jerris.com>
	<CAEorZykBBssx9tsUi3VFqHg2Qh6U9kLBVnfVP2_Ljc=jJ4EDFQ@mail.gmail.com>
	<CAEorZynxUgGOqugfr3O=UN3EgGWjvojMckM=6kzzCFWKFTu8kw@mail.gmail.com>
	<CAEorZymaEWTTeH-4xnGiYF91poq_RXWzHNXUbeX9A+xXwkW+3A@mail.gmail.com>
	<CAKbxfG89-1kPhQh-Y6YAx8GYh7_udTsZxyqbuYSsCaadmprqHQ@mail.gmail.com>
	<CAEorZykV2hMkmQdEJ5-wDjZ3yg1H8rKyKPnWkeCyfwjbk2_f6g@mail.gmail.com>
	<CAKbxfG-Yi+YqbSgJ9_9AF_BLxncivUyeY4ggVygidtSHMrnmDQ@mail.gmail.com>
	<CAEorZyn3HAXHka+_afUyDEuPg1xaSN=VC9xs89hyE=+UMGPVWw@mail.gmail.com>
Message-ID: <CAEorZyko3LcHNZtrMSGrxjPtX5FkRkppqX9zkW+6aQaoCEiwCw@mail.gmail.com>

Take this lua script for instance for testing this out.

-- getting arguments
local confName = argv[1];
local url = argv[2];

-- initialize the API
api = freeswitch.API();

-- download the slide
local cachedPath = api:executeString("http_get " .. url);

-- display the cached path
freeswitch.consoleLog("info", "cached path: " .. cachedPath);

-- display the cached file
api:executeString("conference " .. confName .. " play " .. cachedPath);

On Thu, Jun 23, 2016 at 5:08 PM Danny Gershman <danny.gershman at gmail.com>
wrote:

> I also noticed some weirdness when combining this with mod_http_cache.
>
> On Thu, Jun 23, 2016 at 5:05 PM Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> https://freeswitch.org/jira/browse/FS-9296
>>
>> Update to master and it should work now.
>>
>>
>> On Thu, Jun 23, 2016 at 3:12 PM, Danny Gershman <danny.gershman at gmail.com
>> > wrote:
>>
>>> I am able to see the same on my side, video file is playing now.  Thanks.
>>>
>>> On Thu, Jun 23, 2016 at 4:07 PM Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> Those details are relevant.
>>>>
>>>> I tried it and i see the same problem but only when using the http url.
>>>> If I wget the file then it place so we may have a small bug with
>>>> playing mp4 using htttapi.  I have never tried this before.
>>>>
>>>>
>>>> On Thu, Jun 23, 2016 at 2:01 PM, Danny Gershman <
>>>> danny.gershman at gmail.com> wrote:
>>>>
>>>>> I got this working on a video recorded by FS, but apparently this
>>>>> particular video I was testing with didn't work too well.  Is there a
>>>>> limitation on the videos that can be played back?  Some setting in the
>>>>> conference.conf.xml?
>>>>>
>>>>> On Thu, Jun 23, 2016 at 1:59 PM Danny Gershman <
>>>>> danny.gershman at gmail.com> wrote:
>>>>>
>>>>>> Here is the output of the command I'm executing:
>>>>>>
>>>>>> freeswitch at freeswitch>conference ~video2~conf_123 play
>>>>>> http://download.openbricks.org/sample/H264/h264_sintel_trailer-1080p.mp4
>>>>>> (play) Playing file
>>>>>> http://download.openbricks.org/sample/H264/h264_sintel_trailer-1080p.mp4
>>>>>> 2016-06-23 04:37:23.278786 [DEBUG] avformat.c:1694 sample rate:
>>>>>> 48000, channels: 2
>>>>>> 2016-06-23 04:37:23.450105 [INFO] avformat.c:1303 file
>>>>>> /usr/local/freeswitch/storage/http_file_cache/43772c747785c73f5beb03cc76d617de.mp4
>>>>>> is seekable
>>>>>> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from
>>>>>> '/usr/local/freeswitch/storage/http_file_cache/43772c747785c73f5beb03cc76d617de.mp4':
>>>>>>   Metadata:
>>>>>>     major_brand     : isom
>>>>>>     minor_version   : 512
>>>>>>     compatible_brands: isomiso2avc1mp41
>>>>>>     creation_time   : 1970-01-01 00:00:00
>>>>>>     title           : Sintel Trailer
>>>>>>     artist          : Durian Open Movie Team
>>>>>>     encoder         : Lavf52.62.0
>>>>>>     copyright       : (c) copyright Blender Foundation |
>>>>>> durian.blender.org
>>>>>>     description     : Trailer for the Sintel open movie project
>>>>>>   Duration:
>>>>>> 00:00:52.20
>>>>>> , start:
>>>>>> 0.000000
>>>>>> , bitrate:
>>>>>> 2240 kb/s
>>>>>>     Stream #0.0
>>>>>> (und)
>>>>>> : Video: h264 (High), yuv420p, 1920x1080, 2108 kb/s
>>>>>> , 24 fps
>>>>>> , 24 tbn
>>>>>> , 48 tbc
>>>>>>  (default)
>>>>>>     Metadata:
>>>>>>       creation_time   : 1970-01-01 00:00:00
>>>>>>     Stream #0.1
>>>>>> (und)
>>>>>> : Audio: aac, 48000 Hz, stereo, fltp, 126 kb/s
>>>>>>  (default)
>>>>>>     Metadata:
>>>>>> freeswitch at freeswitch>
>>>>>>       creation_time   : 1970-01-01 00:00:00
>>>>>>
>>>>>> On Thu, Jun 23, 2016 at 1:55 PM Danny Gershman <
>>>>>> danny.gershman at gmail.com> wrote:
>>>>>>
>>>>>>> Other features of mod_av are working fine.  For example, I am to
>>>>>>> record to an rtmp endpoint and see video.  Video codecs are negotiating
>>>>>>> properly.
>>>>>>>
>>>>>>> On Thu, Jun 23, 2016 at 1:54 PM Michael Jerris <mike at jerris.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> I would confirm for sure thats installed, then look at the debug
>>>>>>>> and confirm its actually negotiating a video codec.  Its really hard to
>>>>>>>> make guesses with no information to go on so we are just making wild
>>>>>>>> guesses here.
>>>>>>>>
>>>>>>>> On Jun 23, 2016, at 1:43 PM, Danny Gershman <
>>>>>>>> danny.gershman at gmail.com> wrote:
>>>>>>>>
>>>>>>>> Already have it there, FS was compiled with it.
>>>>>>>>
>>>>>>>> On Thu, Jun 23, 2016 at 1:29 PM Anthony Minessale <
>>>>>>>> anthony.minessale at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> Like mike said, make sure you have main contrib non-free for you
>>>>>>>>> repos and
>>>>>>>>>
>>>>>>>>> libavcodec-extra
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Thu, Jun 23, 2016 at 12:12 PM, Danny Gershman <
>>>>>>>>> danny.gershman at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> I'm trying on 1.6.9 and I'm still not getting results.  Is there
>>>>>>>>>> some specific condition in which this will work.  If I have muxing turned
>>>>>>>>>> on, how does that display back to a softphone?  Should this be played in a
>>>>>>>>>> specific channel?  Is playing a video file to all users in a conference,
>>>>>>>>>> what would the expected response be?
>>>>>>>>>>
>>>>>>>>>> On Thu, Jun 16, 2016 at 8:08 PM Michael Jerris <mike at jerris.com>
>>>>>>>>>> wrote:
>>>>>>>>>>
>>>>>>>>>>> Try 1.6.9 as thats now released, if thats not working, take a
>>>>>>>>>>> look at the debug logs and see what it says.  One common issue is playing
>>>>>>>>>>> mp4 files requires the libav-extras package on debian.
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> On Jun 16, 2016, at 6:54 PM, Danny Gershman <
>>>>>>>>>>> danny.gershman at gmail.com> wrote:
>>>>>>>>>>>
>>>>>>>>>>> I'm having an issue playing back video in a conference, running
>>>>>>>>>>> (1.6.8 tag and loaded up mod_av), which it only plays audio back (no
>>>>>>>>>>> visual).  I'm dialed in using a softphone with VP8 codec.
>>>>>>>>>>>
>>>>>>>>>>> I run command:
>>>>>>>>>>>
>>>>>>>>>>> conference <conf> play http://example.org/play.mp4
>>>>>>>>>>>
>>>>>>>>>>> Any pointers.
>>>>>>>>>>>
>>>>>>>>>>> Thanks,
>>>>>>>>>>> Danny Gershman
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>>>>
>>>> ? http://freeswitch.org/  ? http://cluecon.com/  ?
>>>> http://twitter.com/FreeSWITCH
>>>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
>>>> <http://freeswitch.org/g+>*
>>>>
>>>> ClueCon Weekly Development Call
>>>> ? sip:888 at conference.freeswitch.org  ? +19193869900
>>>>
>>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>>
>> ? http://freeswitch.org/  ? http://cluecon.com/  ?
>> http://twitter.com/FreeSWITCH
>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
>> <http://freeswitch.org/g+>*
>>
>> ClueCon Weekly Development Call
>> ? sip:888 at conference.freeswitch.org  ? +19193869900
>>
>> https://www.youtube.com/watch?v=9XXgW34t40s
>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Fri Jun 24 01:11:32 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 23 Jun 2016 16:11:32 -0500
Subject: [Freeswitch-users] conference play video
In-Reply-To: <CAEorZyko3LcHNZtrMSGrxjPtX5FkRkppqX9zkW+6aQaoCEiwCw@mail.gmail.com>
References: <CAEorZymOmq5b88qBgvCEO5-eoucLx99Si-cYu4HNKi4-Su16fg@mail.gmail.com>
	<618BE480-F3CD-4903-85A5-923FD12C2941@jerris.com>
	<CAEorZy=VFyVvfx+dCCfuBZvR6sS1tGgBfjnTZ_CUj=vj4NS0cA@mail.gmail.com>
	<CAKbxfG-_ouQdkOra2vkPgsoOpkPF6BmuHYzfDMN1BnQSmJfBSg@mail.gmail.com>
	<CAEorZykBsfBcS2-MMDdCeJak+ra3Oy2cSjm64Us+txGeyQhadQ@mail.gmail.com>
	<B319A4CC-3F70-4B58-93B2-556A291F3D27@jerris.com>
	<CAEorZykBBssx9tsUi3VFqHg2Qh6U9kLBVnfVP2_Ljc=jJ4EDFQ@mail.gmail.com>
	<CAEorZynxUgGOqugfr3O=UN3EgGWjvojMckM=6kzzCFWKFTu8kw@mail.gmail.com>
	<CAEorZymaEWTTeH-4xnGiYF91poq_RXWzHNXUbeX9A+xXwkW+3A@mail.gmail.com>
	<CAKbxfG89-1kPhQh-Y6YAx8GYh7_udTsZxyqbuYSsCaadmprqHQ@mail.gmail.com>
	<CAEorZykV2hMkmQdEJ5-wDjZ3yg1H8rKyKPnWkeCyfwjbk2_f6g@mail.gmail.com>
	<CAKbxfG-Yi+YqbSgJ9_9AF_BLxncivUyeY4ggVygidtSHMrnmDQ@mail.gmail.com>
	<CAEorZyn3HAXHka+_afUyDEuPg1xaSN=VC9xs89hyE=+UMGPVWw@mail.gmail.com>
	<CAEorZyko3LcHNZtrMSGrxjPtX5FkRkppqX9zkW+6aQaoCEiwCw@mail.gmail.com>
Message-ID: <CAKbxfG-31gH03UDcF1qoSCZBQTn3GukqbtfsW7tZF=GPcT83wA@mail.gmail.com>

http cache is chis rienzo territory If you open a jira he should do it.


On Thu, Jun 23, 2016 at 4:09 PM, Danny Gershman <danny.gershman at gmail.com>
wrote:

> Take this lua script for instance for testing this out.
>
> -- getting arguments
> local confName = argv[1];
> local url = argv[2];
>
> -- initialize the API
> api = freeswitch.API();
>
> -- download the slide
> local cachedPath = api:executeString("http_get " .. url);
>
> -- display the cached path
> freeswitch.consoleLog("info", "cached path: " .. cachedPath);
>
> -- display the cached file
> api:executeString("conference " .. confName .. " play " .. cachedPath);
>
> On Thu, Jun 23, 2016 at 5:08 PM Danny Gershman <danny.gershman at gmail.com>
> wrote:
>
>> I also noticed some weirdness when combining this with mod_http_cache.
>>
>> On Thu, Jun 23, 2016 at 5:05 PM Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> https://freeswitch.org/jira/browse/FS-9296
>>>
>>> Update to master and it should work now.
>>>
>>>
>>> On Thu, Jun 23, 2016 at 3:12 PM, Danny Gershman <
>>> danny.gershman at gmail.com> wrote:
>>>
>>>> I am able to see the same on my side, video file is playing now.
>>>> Thanks.
>>>>
>>>> On Thu, Jun 23, 2016 at 4:07 PM Anthony Minessale <
>>>> anthony.minessale at gmail.com> wrote:
>>>>
>>>>> Those details are relevant.
>>>>>
>>>>> I tried it and i see the same problem but only when using the http url.
>>>>> If I wget the file then it place so we may have a small bug with
>>>>> playing mp4 using htttapi.  I have never tried this before.
>>>>>
>>>>>
>>>>> On Thu, Jun 23, 2016 at 2:01 PM, Danny Gershman <
>>>>> danny.gershman at gmail.com> wrote:
>>>>>
>>>>>> I got this working on a video recorded by FS, but apparently this
>>>>>> particular video I was testing with didn't work too well.  Is there a
>>>>>> limitation on the videos that can be played back?  Some setting in the
>>>>>> conference.conf.xml?
>>>>>>
>>>>>> On Thu, Jun 23, 2016 at 1:59 PM Danny Gershman <
>>>>>> danny.gershman at gmail.com> wrote:
>>>>>>
>>>>>>> Here is the output of the command I'm executing:
>>>>>>>
>>>>>>> freeswitch at freeswitch>conference ~video2~conf_123 play
>>>>>>> http://download.openbricks.org/sample/H264/h264_sintel_trailer-1080p.mp4
>>>>>>> (play) Playing file
>>>>>>> http://download.openbricks.org/sample/H264/h264_sintel_trailer-1080p.mp4
>>>>>>> 2016-06-23 04:37:23.278786 [DEBUG] avformat.c:1694 sample rate:
>>>>>>> 48000, channels: 2
>>>>>>> 2016-06-23 04:37:23.450105 [INFO] avformat.c:1303 file
>>>>>>> /usr/local/freeswitch/storage/http_file_cache/43772c747785c73f5beb03cc76d617de.mp4
>>>>>>> is seekable
>>>>>>> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from
>>>>>>> '/usr/local/freeswitch/storage/http_file_cache/43772c747785c73f5beb03cc76d617de.mp4':
>>>>>>>   Metadata:
>>>>>>>     major_brand     : isom
>>>>>>>     minor_version   : 512
>>>>>>>     compatible_brands: isomiso2avc1mp41
>>>>>>>     creation_time   : 1970-01-01 00:00:00
>>>>>>>     title           : Sintel Trailer
>>>>>>>     artist          : Durian Open Movie Team
>>>>>>>     encoder         : Lavf52.62.0
>>>>>>>     copyright       : (c) copyright Blender Foundation |
>>>>>>> durian.blender.org
>>>>>>>     description     : Trailer for the Sintel open movie project
>>>>>>>   Duration:
>>>>>>> 00:00:52.20
>>>>>>> , start:
>>>>>>> 0.000000
>>>>>>> , bitrate:
>>>>>>> 2240 kb/s
>>>>>>>     Stream #0.0
>>>>>>> (und)
>>>>>>> : Video: h264 (High), yuv420p, 1920x1080, 2108 kb/s
>>>>>>> , 24 fps
>>>>>>> , 24 tbn
>>>>>>> , 48 tbc
>>>>>>>  (default)
>>>>>>>     Metadata:
>>>>>>>       creation_time   : 1970-01-01 00:00:00
>>>>>>>     Stream #0.1
>>>>>>> (und)
>>>>>>> : Audio: aac, 48000 Hz, stereo, fltp, 126 kb/s
>>>>>>>  (default)
>>>>>>>     Metadata:
>>>>>>> freeswitch at freeswitch>
>>>>>>>       creation_time   : 1970-01-01 00:00:00
>>>>>>>
>>>>>>> On Thu, Jun 23, 2016 at 1:55 PM Danny Gershman <
>>>>>>> danny.gershman at gmail.com> wrote:
>>>>>>>
>>>>>>>> Other features of mod_av are working fine.  For example, I am to
>>>>>>>> record to an rtmp endpoint and see video.  Video codecs are negotiating
>>>>>>>> properly.
>>>>>>>>
>>>>>>>> On Thu, Jun 23, 2016 at 1:54 PM Michael Jerris <mike at jerris.com>
>>>>>>>> wrote:
>>>>>>>>
>>>>>>>>> I would confirm for sure thats installed, then look at the debug
>>>>>>>>> and confirm its actually negotiating a video codec.  Its really hard to
>>>>>>>>> make guesses with no information to go on so we are just making wild
>>>>>>>>> guesses here.
>>>>>>>>>
>>>>>>>>> On Jun 23, 2016, at 1:43 PM, Danny Gershman <
>>>>>>>>> danny.gershman at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>> Already have it there, FS was compiled with it.
>>>>>>>>>
>>>>>>>>> On Thu, Jun 23, 2016 at 1:29 PM Anthony Minessale <
>>>>>>>>> anthony.minessale at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> Like mike said, make sure you have main contrib non-free for you
>>>>>>>>>> repos and
>>>>>>>>>>
>>>>>>>>>> libavcodec-extra
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Thu, Jun 23, 2016 at 12:12 PM, Danny Gershman <
>>>>>>>>>> danny.gershman at gmail.com> wrote:
>>>>>>>>>>
>>>>>>>>>>> I'm trying on 1.6.9 and I'm still not getting results.  Is there
>>>>>>>>>>> some specific condition in which this will work.  If I have muxing turned
>>>>>>>>>>> on, how does that display back to a softphone?  Should this be played in a
>>>>>>>>>>> specific channel?  Is playing a video file to all users in a conference,
>>>>>>>>>>> what would the expected response be?
>>>>>>>>>>>
>>>>>>>>>>> On Thu, Jun 16, 2016 at 8:08 PM Michael Jerris <mike at jerris.com>
>>>>>>>>>>> wrote:
>>>>>>>>>>>
>>>>>>>>>>>> Try 1.6.9 as thats now released, if thats not working, take a
>>>>>>>>>>>> look at the debug logs and see what it says.  One common issue is playing
>>>>>>>>>>>> mp4 files requires the libav-extras package on debian.
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> On Jun 16, 2016, at 6:54 PM, Danny Gershman <
>>>>>>>>>>>> danny.gershman at gmail.com> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>> I'm having an issue playing back video in a conference, running
>>>>>>>>>>>> (1.6.8 tag and loaded up mod_av), which it only plays audio back (no
>>>>>>>>>>>> visual).  I'm dialed in using a softphone with VP8 codec.
>>>>>>>>>>>>
>>>>>>>>>>>> I run command:
>>>>>>>>>>>>
>>>>>>>>>>>> conference <conf> play http://example.org/play.mp4
>>>>>>>>>>>>
>>>>>>>>>>>> Any pointers.
>>>>>>>>>>>>
>>>>>>>>>>>> Thanks,
>>>>>>>>>>>> Danny Gershman
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>> http://www.cluecon.com
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>>>>>
>>>>> ? http://freeswitch.org/  ? http://cluecon.com/  ?
>>>>> http://twitter.com/FreeSWITCH
>>>>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
>>>>> <http://freeswitch.org/g+>*
>>>>>
>>>>> ClueCon Weekly Development Call
>>>>> ? sip:888 at conference.freeswitch.org  ? +19193869900
>>>>>
>>>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>>>
>>> ? http://freeswitch.org/  ? http://cluecon.com/  ?
>>> http://twitter.com/FreeSWITCH
>>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
>>> <http://freeswitch.org/g+>*
>>>
>>> ClueCon Weekly Development Call
>>> ? sip:888 at conference.freeswitch.org  ? +19193869900
>>>
>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From ashwinrkjain at gmail.com  Fri Jun 24 06:53:01 2016
From: ashwinrkjain at gmail.com (Ashwin Jain)
Date: Fri, 24 Jun 2016 08:23:01 +0530
Subject: [Freeswitch-users] Audio skip issue with simultaneous ringing
In-Reply-To: <CAEwTsAg0+rVhJpqiQWePJ5vUv5xN_pyFnc=RGSvE2tcgGjLFFw@mail.gmail.com>
References: <CAM9SNVTQJo5jCZ+jb9Yy6fErb7UdUtH1QpbNn=8kPkdaP=m0vg@mail.gmail.com>
	<CAEwTsAjozSr2eQyPht1nxP5AophRUeqgxYaWtqBrn5+HA9o1zw@mail.gmail.com>
	<CAM9SNVSQSNp7gTfCV2gJ6t8BYmVeKFCP01dC9WN+b0-RENMMAg@mail.gmail.com>
	<CAM9SNVTXTNit_MwrDsT1TZPDeNoY=rEN=2rqhjEKNSkqjQADJg@mail.gmail.com>
	<CAEwTsAg0+rVhJpqiQWePJ5vUv5xN_pyFnc=RGSvE2tcgGjLFFw@mail.gmail.com>
Message-ID: <CAM9SNVSH=KJncd=NzVKGcbM0yartYwtouSRwcYqJNkFotHM+iw@mail.gmail.com>

Sure. Thanks. I will do that and get back to you.

On Thu, Jun 23, 2016 at 8:56 PM, Michael Jerris <mike at jerris.com> wrote:

> My suggestion is for you to confirm this by testing the latest release.
> There have been over 1000 bug fixes since the version you are using, I
> don't recall if this specific issue is included in those or not.
>
>
> On Thursday, June 23, 2016, Ashwin Jain <ashwinrkjain at gmail.com> wrote:
>
>> Hi Michael,
>>
>> Can you please confirm that it is the freeswitch issue and not our
>> configuration issue? That would be of really great help.
>>
>> On Tue, Jun 21, 2016 at 1:46 AM, Ashwin Jain <ashwinrkjain at gmail.com>
>> wrote:
>>
>>> Sure. We can update to the latest version. Can you confirm if this is
>>> the issue, or is it some setup/configuration issue with our system?
>>>
>>> On Mon, Jun 20, 2016 at 9:04 PM, Michael Jerris <mike at jerris.com> wrote:
>>>
>>>> You are using older versions, I would first check if we have fixed this
>>>> by testing the latest releases.  1.4 is only receiving critical security
>>>> releases now and would not receive a fix if we found a problem.
>>>>
>>>> On Saturday, June 18, 2016, Ashwin Jain <ashwinrkjain at gmail.com> wrote:
>>>>
>>>>> Hi all,
>>>>>
>>>>> We are running Freeswitch version 1.4.26. We are facing a major
>>>>> problem related to audio skip with simultaneous ringing. In one flow we are
>>>>> ringing around 5 agents at the same time. Whenever agent 1 (any one of the
>>>>> agent) picks up the call, other legs (to other agents) gets a hangup event.
>>>>> Due to this, hangup event to all the other agent "CALL_ANSWER" event on the
>>>>> other side gets delayed by around 2-3 seconds and generally caller missed
>>>>> the first word from the agent.
>>>>>
>>>>> Event 1: Call Answered by agent 1
>>>>> Event 2: Hangup for agent 2 (due to LOSE_RACE)
>>>>> Event 3: Hangup for agent 3 (due to LOSE_RACE)
>>>>> Event 4: Hangup for agent 4 (due to LOSE_RACE)
>>>>> Event 5: Hangup for agent 5 (due to LOSE_RACE)
>>>>> Event 6: Call answered to caller
>>>>>
>>>>> Time different between Event 1 and Event 6 is about 2-3 seconds.
>>>>> Any help would be really really helpful
>>>>>
>>>>> --
>>>>> Thanks and Regards,
>>>>> Ashwin Jain
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>>
>>> *Ashwin Jain*
>>>
>>
>>
>> --
>>
>> *Ashwin Jain*
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Ashwin Jain* / Director of Engineering

*Email** :* ashwin at metroguild.com <chiqeetaj at metroguild.com>

*Phone :* (415) 800-0834 / +91-97424-94330
*Website* : www.metroguild.com
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From ssinyagin at gmail.com  Fri Jun 24 09:30:39 2016
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Fri, 24 Jun 2016 07:30:39 +0200
Subject: [Freeswitch-users] Debian 8 package i386?
In-Reply-To: <04e201d1cd78$5b890110$129b0330$@secureachsystems.com>
References: <04e201d1cd78$5b890110$129b0330$@secureachsystems.com>
Message-ID: <CACOYK=gJH1ZTpiPT96sfwNOnOm5bdQNOn0GNKGgZxvYhxFnDrQ@mail.gmail.com>

Jason, this should work:
https://freeswitch.org/confluence/display/FREESWITCH/Building+debs+for+other+architectures

I'm using it for ARM architecture, but i386 should work as well. Let
me know if something's failing.



On Thu, Jun 23, 2016 at 7:54 PM, Jason Moran
<jmoran at secureachsystems.com> wrote:
> I have VMs of Deb8 for both 64bit and 32bit/i386.
> The instructions work great for 64bit, and FS installed as expected.
>
>
>
> However, on i386 the command
>
> # apt-get update && apt-get install -y freeswitch-meta-all
>
> returns
>
> E: Unable to locate package freeswitch-meta-all
>
>
>
> I looked in
> http://files.freeswitch.org/repo/deb/freeswitch-1.6/db/packages.db and see
> i386 packages for 1.4 (Wheezy) but for 1.6 I only see amd64.
>
> Is our old x86 server unsupported now?
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From kbdfck at gmail.com  Fri Jun 24 10:24:26 2016
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Fri, 24 Jun 2016 09:24:26 +0300
Subject: [Freeswitch-users] Short ringback file duration leads to bridge
	cancel with LOSE_RACE
Message-ID: <CA+GmxbaWO2=JLMJoMjmnq8tWgHW-Wi41N=ToSLf9WS2AQMnp=Q@mail.gmail.com>

Hi all!

When using ringback="http://something" or ringback="file.wav" and
file duration is shorter than bridge timeout, FS doesn't repeat it
from beginning. Instead, bridge cancels with LOSE_RACE despite there
is only one outbound channel in bridge.
Seems it doesn't depend on early media settings on bridge.

Does anybody know if this is intended behavior or should I file a jira?



-- 
Best regards,

Dmitry Sytchev,
IT Engineer


From jonlederman at gmail.com  Fri Jun 24 16:37:45 2016
From: jonlederman at gmail.com (Jon Lederman)
Date: Fri, 24 Jun 2016 08:37:45 -0400
Subject: [Freeswitch-users] Codec Reset Issue With Media Bug Invoked On Both
	Call Legs
Message-ID: <021848EC-E397-4EF7-8073-E7CD6F2C9EFA@gmail.com>

Hi,

We are utilizing a module that performs processing on audio using a media bug.  We are finding that if the media bug is invoked on only one call leg, we are seeing very frequent codec resets on the call leg that doesn?t have the media bug invoked leading to very choppy audio.  If the media bug is on both call legs we don?t see the issue or the number of codec resets is much lower.  

Any thoughts on what may be causing this and why having the media bug on both call legs would make a difference?


Thanks.

-Jon

From christophe.yann at gmail.com  Fri Jun 24 13:07:10 2016
From: christophe.yann at gmail.com (yann christophe)
Date: Fri, 24 Jun 2016 11:07:10 +0200
Subject: [Freeswitch-users] Freeswitch on AWS
Message-ID: <CA+qsECtiQ6scQLc-htQngbqprfdavWdc7QQ3OLEC2Bujy0pXVg@mail.gmail.com>

Hello everyone,

I would like to know if someone has an experience on AWS with Freeswitch.

Because i'll deploy 2 Freeswitch  servers on EC2 and i would like to
use S3 for the directory "Dialplan". I want this Directory is the same
for these 2 servers,  I currently use rsync, maybe it's better to use
S3 for that.

What do you suggest me ? I just started to work on AWS

Thanks
Best regards


From olegstolyar at gmail.com  Fri Jun 24 17:21:56 2016
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Fri, 24 Jun 2016 06:21:56 -0700
Subject: [Freeswitch-users] Freeswitch on AWS
In-Reply-To: <CA+qsECtiQ6scQLc-htQngbqprfdavWdc7QQ3OLEC2Bujy0pXVg@mail.gmail.com>
References: <CA+qsECtiQ6scQLc-htQngbqprfdavWdc7QQ3OLEC2Bujy0pXVg@mail.gmail.com>
Message-ID: <CAC8JMOJFLXJHEFyEcSAB-2nY6pLuAKWziyJQTFFGukOun5j6bg@mail.gmail.com>

I regularly transfer large files between my FS servers in EC2 and S3 using
AWS s3 command line utility (
http://docs.aws.amazon.com/cli/latest/reference/s3/index.html).

The sync operation called from a cron job is what you are probably looking
for to replace rsync.


On Fri, Jun 24, 2016 at 2:07 AM, yann christophe <christophe.yann at gmail.com>
wrote:

> Hello everyone,
>
> I would like to know if someone has an experience on AWS with Freeswitch.
>
> Because i'll deploy 2 Freeswitch  servers on EC2 and i would like to
> use S3 for the directory "Dialplan". I want this Directory is the same
> for these 2 servers,  I currently use rsync, maybe it's better to use
> S3 for that.
>
> What do you suggest me ? I just started to work on AWS
>
> Thanks
> Best regards
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Fri Jun 24 22:14:38 2016
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 24 Jun 2016 19:14:38 +0100
Subject: [Freeswitch-users] Question regarding UUID stacking in a FIFO
In-Reply-To: <64F76782-7F99-40AD-9134-7B1A2340B722@gmail.com>
References: <64F76782-7F99-40AD-9134-7B1A2340B722@gmail.com>
Message-ID: <CAFiqYukypQ+B-_5cR2XHoVAXkkx=A4=r8wx1JVSfGw+m2My8hQ@mail.gmail.com>

Since realm+key are a unique pair storing a single value you can't do it
with mod_db. Anything build around fetching,editing,storing the value will
be subject to race conditions if there are 2 concurrent calls accessing it.

You could implement this yourself using a custom transactional database and
accessing it in Lua via freeswitch.Dbh. Begin a transaction, select the
first UUID ordered by perhaps insertion uepoch, delete that uuid, commit,
then use that uuid. Using SELECT FOR UPDATE will prevent another concurrent
call getting the same uuid - the other will block until the commit at which
point it's no longer in the table. That blocking should be fine because
it's a very short transaction. Just insert another row to push a new uuid
onto the stack.

On 23 June 2016 at 15:35, David Ponzone <david.ponzone at gmail.com> wrote:

> Guys,
>
> I am searching for a way to stack UUID into a FIFO.
> What I need is something like db insert/realm/key/${uuid} or hash
> insert/realm/key/${uuid} but that I need to keep several uuids with the
> same realm/key tuple.
>
> The ideal would be:
> I first insert realm/key/${uuid1}
> then insert real/ley/{uuid2}
>
> then if I do select/realm/key, it pops out ${uuid1)
> I can then manually remove it, and the next select will pop out ${uuid2}
>
> I don?t think it?s possible with hash/db, so does anyone a way/trick to
> achieve that ?
>
> (the goal is to intercept several simultaneous incoming calls in the
> RINGING state)
>
> Thank you
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gled at remote-shell.net  Fri Jun 24 22:16:54 2016
From: gled at remote-shell.net (=?UTF-8?Q?Tristan_Mah=c3=a9?=)
Date: Fri, 24 Jun 2016 11:16:54 -0700
Subject: [Freeswitch-users] Freeswitch on AWS
In-Reply-To: <CA+qsECtiQ6scQLc-htQngbqprfdavWdc7QQ3OLEC2Bujy0pXVg@mail.gmail.com>
References: <CA+qsECtiQ6scQLc-htQngbqprfdavWdc7QQ3OLEC2Bujy0pXVg@mail.gmail.com>
Message-ID: <8ba010fd-f8ae-7324-3432-2d842057b6c1@remote-shell.net>

Hi,

It could work, as the dialplan is only loaded once or on demand. We are
using AWS with success here.

One other thing you may want to be aware of, is to use high cpu
instances, to avoid having some weird artifact due to noisy neighboors
on the physical host and steal time going to the roof, plus the usual timing

consistency...

Best,

Gled.

On 06/24/2016 02:07 AM, yann christophe wrote:
> Hello everyone,
>
> I would like to know if someone has an experience on AWS with Freeswitch.
>
> Because i'll deploy 2 Freeswitch  servers on EC2 and i would like to
> use S3 for the directory "Dialplan". I want this Directory is the same
> for these 2 servers,  I currently use rsync, maybe it's better to use
> S3 for that.
>
> What do you suggest me ? I just started to work on AWS
>
> Thanks
> Best regards
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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From anthony.minessale at gmail.com  Sat Jun 25 02:34:41 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 24 Jun 2016 17:34:41 -0500
Subject: [Freeswitch-users] Codec Reset Issue With Media Bug Invoked On
 Both Call Legs
In-Reply-To: <021848EC-E397-4EF7-8073-E7CD6F2C9EFA@gmail.com>
References: <021848EC-E397-4EF7-8073-E7CD6F2C9EFA@gmail.com>
Message-ID: <CAKbxfG_=mDhhbnEMPQSFi4pirQkAdJ6T8tjPC48ZNB2=YiVh4Q@mail.gmail.com>

What is a codec reset?

On Friday, June 24, 2016, Jon Lederman <jonlederman at gmail.com> wrote:

> Hi,
>
> We are utilizing a module that performs processing on audio using a media
> bug.  We are finding that if the media bug is invoked on only one call leg,
> we are seeing very frequent codec resets on the call leg that doesn?t have
> the media bug invoked leading to very choppy audio.  If the media bug is on
> both call legs we don?t see the issue or the number of codec resets is much
> lower.
>
> Any thoughts on what may be causing this and why having the media bug on
> both call legs would make a difference?
>
>
> Thanks.
>
> -Jon
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <javascript:;>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From jonlederman at gmail.com  Sat Jun 25 02:45:22 2016
From: jonlederman at gmail.com (Jon Lederman)
Date: Fri, 24 Jun 2016 18:45:22 -0400
Subject: [Freeswitch-users] Codec Reset Issue With Media Bug Invoked On
	Both Call Legs
In-Reply-To: <CAKbxfG_=mDhhbnEMPQSFi4pirQkAdJ6T8tjPC48ZNB2=YiVh4Q@mail.gmail.com>
References: <021848EC-E397-4EF7-8073-E7CD6F2C9EFA@gmail.com>
	<CAKbxfG_=mDhhbnEMPQSFi4pirQkAdJ6T8tjPC48ZNB2=YiVh4Q@mail.gmail.com>
Message-ID: <94837390-55C1-49CF-8FEF-B8D288AE2914@gmail.com>

Hi,


By codec reset I mean the following:

2016-05-04 03:51:32.464652 [DEBUG] mod_opus.c:661 Opus decoder stats: Frames[0] PLC[0] FEC[0]
2016-05-04 03:51:32.464652 [DEBUG] mod_opus.c:559 Opus encoder: set bitrate to local settings [72000bps]

Thanks

Jon


Sent from my iPhone

> On Jun 24, 2016, at 6:34 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> 
> What is a codec reset?
> 
>> On Friday, June 24, 2016, Jon Lederman <jonlederman at gmail.com> wrote:
>> Hi,
>> 
>> We are utilizing a module that performs processing on audio using a media bug.  We are finding that if the media bug is invoked on only one call leg, we are seeing very frequent codec resets on the call leg that doesn?t have the media bug invoked leading to very choppy audio.  If the media bug is on both call legs we don?t see the issue or the number of codec resets is much lower.
>> 
>> Any thoughts on what may be causing this and why having the media bug on both call legs would make a difference?
>> 
>> 
>> Thanks.
>> 
>> -Jon
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> -- 
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
> 
> ? http://freeswitch.org/  ? http://cluecon.com/  ? http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? http://freeswitch.org/g+
> 
> ClueCon Weekly Development Call 
> ? sip:888 at conference.freeswitch.org  ? +19193869900 
> 
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From david at forestit.co.uk  Sat Jun 25 02:21:59 2016
From: david at forestit.co.uk (David Mitchell)
Date: Fri, 24 Jun 2016 22:21:59 +0000
Subject: [Freeswitch-users] BLF for Sip gateway?
Message-ID: <VI1PR0301MB19345656B100669AE82D0377922E0@VI1PR0301MB1934.eurprd03.prod.outlook.com>

I am setting up a small PBX with freeswitch / debian etc

I am using Linksys SP3000 ATA to interface with the PSTN.

In/Outbound calls work.

I would like to set up a BLF to show if the PST N line is in use, I can configure it on the phone (Yealink T23G) and associate it with the PSTN extension (191) and if I dial 191 from another extension the lamp shows busy (and I am connected to the PSTN line...)

However it doesn't show busy for inbound PSTN calls, nor for calls routed to the PSTN gateway via the dialplan (using action = bridge to <pstn number>@<SP200 ip address>:5061) - the calls do get connected to the PSTN.

Is there a way to make this work - perhaps by setting the presence within the dialplan actions?

David




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From yu at yu-boot.ru  Sat Jun 25 03:00:28 2016
From: yu at yu-boot.ru (Yu Boot)
Date: Sat, 25 Jun 2016 02:00:28 +0300
Subject: [Freeswitch-users] Routing inbound calls from gateways?
In-Reply-To: <94837390-55C1-49CF-8FEF-B8D288AE2914@gmail.com>
References: <021848EC-E397-4EF7-8073-E7CD6F2C9EFA@gmail.com>
	<CAKbxfG_=mDhhbnEMPQSFi4pirQkAdJ6T8tjPC48ZNB2=YiVh4Q@mail.gmail.com>
	<94837390-55C1-49CF-8FEF-B8D288AE2914@gmail.com>
Message-ID: <a409917f-5d57-a81d-cd8a-e8a575d9106d@yu-boot.ru>

Hello. Asked it sometimes ago, but unfortunately I didn't get any answer.

Is it possible to route inbound calls from the gateways described in SIP 
profile in other way but
             <condition field="network_addr" expression="^1\.2\.3\.4$" 
break="never">
?

context variable in gateway parameters seems to be totally ignored by 
FS, but it works for password-protected users.

Or why it's impossible to just auth that calls in FS by peer name-peer 
IP pair, like Asterisk does?


From krice at freeswitch.org  Sat Jun 25 03:26:44 2016
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 24 Jun 2016 18:26:44 -0500
Subject: [Freeswitch-users] Routing inbound calls from gateways?
In-Reply-To: <a409917f-5d57-a81d-cd8a-e8a575d9106d@yu-boot.ru>
References: <021848EC-E397-4EF7-8073-E7CD6F2C9EFA@gmail.com>
	<CAKbxfG_=mDhhbnEMPQSFi4pirQkAdJ6T8tjPC48ZNB2=YiVh4Q@mail.gmail.com>
	<94837390-55C1-49CF-8FEF-B8D288AE2914@gmail.com>
	<a409917f-5d57-a81d-cd8a-e8a575d9106d@yu-boot.ru>
Message-ID: <7CEEF7B3-A9E0-47DD-A4A9-652D5B81885E@freeswitch.org>

Personally for ip auth i do it in the dialplan either like the network_addr you do below or via xml curl and have a back end return the dialplan for any given ip.

You can also get fancy with the acl api in the condition

Sent from my iPhone

> On Jun 24, 2016, at 6:00 PM, Yu Boot <yu at yu-boot.ru> wrote:
> 
> Hello. Asked it sometimes ago, but unfortunately I didn't get any answer.
> 
> Is it possible to route inbound calls from the gateways described in SIP 
> profile in other way but
>             <condition field="network_addr" expression="^1\.2\.3\.4$" 
> break="never">
> ?
> 
> context variable in gateway parameters seems to be totally ignored by 
> FS, but it works for password-protected users.
> 
> Or why it's impossible to just auth that calls in FS by peer name-peer 
> IP pair, like Asterisk does?
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From deepikay at iiitd.ac.in  Sat Jun 25 15:09:04 2016
From: deepikay at iiitd.ac.in (Deepika Yadav)
Date: Sat, 25 Jun 2016 16:39:04 +0530
Subject: [Freeswitch-users] Early Ring timeout
Message-ID: <CAM99_Y=dQPg8jQu6u2H49XhzF_5qmnLBGQJhMdtR=NThtLz8dQ@mail.gmail.com>

Hi,

I am originating calls to mobile number from ESL APIs and want to capture
the events of switch-off, busy phone, call rejection etc

Corresponding Hangup causes in Freeswitch are -NO_USER_RESPONSE, USER_BUSY,
CALL_REJECTED

My diaplan is :

<extension name="ivr_demo">
      <condition field="destination_number" expression="^4441$">
<action application="playback"
data="/usr/local/freeswitch/recordings/${folder}/trailer/trailer.wav"/>
</condition>
</extension>

command is originate {folder=abc}sofia/gateway/MySIP/91XXXXXXXXXX 4441

The target phone recieves only around six rings and if phone is not
answered, call automatically hangups by reporting hangup cause as
NORMAL_CLEARING


whereas I originate a conference call :

originate {folder=abc}sofia/gateway/MySIP/91XXXXXXXXXX &conference(abc)

the phone get around 12 rings and if call not anwered, hangup cause is
reported as NO_USER_RESPONSE

Is something wrong with the dialplan?

Regards,
Deepika
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From gregor at infomedia.si  Sat Jun 25 16:02:25 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Sat, 25 Jun 2016 14:02:25 +0200
Subject: [Freeswitch-users] Verto + external database
Message-ID: <CAC7ttFEnV1iDf+7qbA9DJgfa+tcAotS9aN4fKRXB9gnYPcsFxg@mail.gmail.com>

I guess that verto uses db for storing user registrations.

I am using external database (MS SQL) instead integrated sqlite, but I
cannot se verto users in registrations table.

Does verto uses external db?
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From yehavi.bourvine at gmail.com  Sat Jun 25 18:27:42 2016
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sat, 25 Jun 2016 17:27:42 +0300
Subject: [Freeswitch-users] tone detect tolerance?
Message-ID: <CAAtWKeESZh5B9ndupmgXP2dtcAxfLoprEhPg7vB8cG_b2Bk+TA@mail.gmail.com>

Hi,

  Lately, probably after updating from 1.6.7 to 1.6.9, people are
complaining that some faxes are received as voicemail. Indeed I don't see
in my logs that the tone detect found the fax CNG tone.
Hence, I have a couple of questions:

- What is the tolerance of tone_detect function?
- If I call it a few times with slightly different frequencies, will all of
them trigger, or only the last one?

            Thanks! __Yehavi:
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From william at williamcollsassoc.ca  Sun Jun 26 17:33:33 2016
From: william at williamcollsassoc.ca (William Colls)
Date: Sun, 26 Jun 2016 09:33:33 -0400
Subject: [Freeswitch-users] Photo
Message-ID: <576FD9AD.70909@williamcollsassoc.ca>


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From mike at jerris.com  Sun Jun 26 18:32:13 2016
From: mike at jerris.com (Michael Jerris)
Date: Sun, 26 Jun 2016 10:32:13 -0400
Subject: [Freeswitch-users] Verto + external database
In-Reply-To: <CAC7ttFEnV1iDf+7qbA9DJgfa+tcAotS9aN4fKRXB9gnYPcsFxg@mail.gmail.com>
References: <CAC7ttFEnV1iDf+7qbA9DJgfa+tcAotS9aN4fKRXB9gnYPcsFxg@mail.gmail.com>
Message-ID: <CAEwTsAiSsqE4ZxYoZL+hX5gtUeK0zgmMk_4KF9YAC9Hk141LVA@mail.gmail.com>

verto stores registrations in memory,
not in db.

On Saturday, June 25, 2016, Gregor Nanger <gregor at infomedia.si> wrote:

> I guess that verto uses db for storing user registrations.
>
> I am using external database (MS SQL) instead integrated sqlite, but I
> cannot se verto users in registrations table.
>
> Does verto uses external db?
>
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From aubalde at presenceco.com  Sun Jun 26 18:47:39 2016
From: aubalde at presenceco.com (=?UTF-8?Q?Agust=C3=AD_Ubalde?=)
Date: Sun, 26 Jun 2016 16:47:39 +0200
Subject: [Freeswitch-users] Changing Auto stun / rtp / DTLS
In-Reply-To: <CAKbxfG8-6W-hT=cEdn=3QBCic8Os7FY1vadVA-r7m4r-6+yo5A@mail.gmail.com>
References: <CAKThY069UjJHRuu9OL0Pi8CMhP=qR8V97RUirEWKwGWFC_8NkA@mail.gmail.com>
	<CAKbxfG8-6W-hT=cEdn=3QBCic8Os7FY1vadVA-r7m4r-6+yo5A@mail.gmail.com>
Message-ID: <CAKThY07KzmOo1f0CJiY6HeXQeQFmztRutZMhicb66C+tiuTjMw@mail.gmail.com>

Hi Anthony,

This could cause that the connection works intermittently? Any idea to
solve the problem?


Thanks!

*PRESENCE TECHNOLOGY*
*Agust? Ubalde Bellot*
Chief Developer
C/ Comte Urgell 240 3A
Barcelona 08036
aubalde at presenceco.com

Ph: +34 93 10 10 300
Fx: +34 93 10 10 333

*www.presenceco.com* <http://www.presenceco.com/>

*Follow us on:*

*[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
<http://www.youtube.com/PresenceTechnology>*[image: in]*
<http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]*
<http://www.slideshare.net/PresenceTech>*[image: fb]*
<http://www.facebook.com/presencetech>

For additional information, please visit our website *www.presenceco.com*
<http://www.presenceco.com/>

2016-06-23 19:16 GMT+02:00 Anthony Minessale <anthony.minessale at gmail.com>:

> It happens when a candidate has not gotten any stun from the expected
> places but it gets some from another place that is not the right address
> but does have the right credentials.
> It will lock on to the remote destination for that candidate instead.
> This is usually encountered when the other side is asymmetric nat.
>
> On Thu, Jun 23, 2016 at 11:20 AM, Agust? Ubalde <aubalde at presenceco.com>
> wrote:
>
>> HI all,
>>
>> In the log file I observed the trace *Changing Auto stun / rtp / DTLS*
>> It only happens on connections from a particular IP. Anyone know explain to
>> that is?
>>
>>
>> Regards,
>>
>> *PRESENCE TECHNOLOGY*
>> *Agust? Ubalde Bellot*
>> Chief Developer
>> C/ Comte Urgell 240 3A
>> Barcelona 08036
>> aubalde at presenceco.com
>>
>> Ph: +34 93 10 10 300
>> Fx: +34 93 10 10 333
>>
>> *www.presenceco.com* <http://www.presenceco.com/>
>>
>> *Follow us on:*
>>
>> *[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
>> <http://www.youtube.com/PresenceTechnology>*[image: in]*
>> <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]*
>> <http://www.slideshare.net/PresenceTech>*[image: fb]*
>> <http://www.facebook.com/presencetech>
>>
>> For additional information, please visit our website *www.presenceco.com*
>> <http://www.presenceco.com/>
>>
>>
>> *Presence Technology - DisclaimerThis message, its content and any file
>> attached thereto is for the intended recipient only and is confidential and
>> /or privileged. If you have received this e-mail in error or had access to
>> it, you should note that the information in it is private and any use
>> thereof is unauthorized. In such an event please notify us by e-mail or by
>> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by
>> whatsoever means and any transmission or dissemination thereof to other
>> persons is prohibited. It should be deleted immediately from your system.
>> Presence Technology reserves the right to take legal action against any
>> persons unlawfully gaining access to the content of any external message it
>> has emitted.*
>>
>> *For additional information, please visit our website **www.presenceco.com
>> <http://www.presenceco.com/>*
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 



*Presence Technology - DisclaimerThis message, its content and any file 
attached thereto is for the intended recipient only and is confidential and 
/or privileged. If you have received this e-mail in error or had access to 
it, you should note that the information in it is private and any use 
thereof is unauthorized. In such an event please notify us by e-mail or by 
telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by 
whatsoever means and any transmission or dissemination thereof to other 
persons is prohibited. It should be deleted immediately from your system. 
Presence Technology reserves the right to take legal action against any 
persons unlawfully gaining access to the content of any external message it 
has emitted.*

*For additional information, please visit our website **www.presenceco.com 
<http://www.presenceco.com/>*

 
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From anthony.minessale at gmail.com  Sun Jun 26 19:05:50 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 26 Jun 2016 10:05:50 -0500
Subject: [Freeswitch-users] Changing Auto stun / rtp / DTLS
In-Reply-To: <CAKThY07KzmOo1f0CJiY6HeXQeQFmztRutZMhicb66C+tiuTjMw@mail.gmail.com>
References: <CAKThY069UjJHRuu9OL0Pi8CMhP=qR8V97RUirEWKwGWFC_8NkA@mail.gmail.com>
	<CAKbxfG8-6W-hT=cEdn=3QBCic8Os7FY1vadVA-r7m4r-6+yo5A@mail.gmail.com>
	<CAKThY07KzmOo1f0CJiY6HeXQeQFmztRutZMhicb66C+tiuTjMw@mail.gmail.com>
Message-ID: <CAKbxfG__kyzdSc602vFCGUU7aS9NRPU7RfwpHKM0HzJ1GvxXaw@mail.gmail.com>

You didnt describe any problem really other than one host prints that
message.  The bes solution is to enable symetric nat for that host.



On Sunday, June 26, 2016, Agust? Ubalde <aubalde at presenceco.com> wrote:

> Hi Anthony,
>
> This could cause that the connection works intermittently? Any idea to
> solve the problem?
>
>
> Thanks!
>
> *PRESENCE TECHNOLOGY*
> *Agust? Ubalde Bellot*
> Chief Developer
> C/ Comte Urgell 240 3A
> Barcelona 08036
> aubalde at presenceco.com
> <javascript:_e(%7B%7D,'cvml','aubalde at presenceco.com');>
>
> Ph: +34 93 10 10 300
> Fx: +34 93 10 10 333
>
> *www.presenceco.com* <http://www.presenceco.com/>
>
> *Follow us on:*
>
> *[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
> <http://www.youtube.com/PresenceTechnology>*[image: in]*
> <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]*
> <http://www.slideshare.net/PresenceTech>*[image: fb]*
> <http://www.facebook.com/presencetech>
>
> For additional information, please visit our website *www.presenceco.com*
> <http://www.presenceco.com/>
>
> 2016-06-23 19:16 GMT+02:00 Anthony Minessale <anthony.minessale at gmail.com
> <javascript:_e(%7B%7D,'cvml','anthony.minessale at gmail.com');>>:
>
>> It happens when a candidate has not gotten any stun from the expected
>> places but it gets some from another place that is not the right address
>> but does have the right credentials.
>> It will lock on to the remote destination for that candidate instead.
>> This is usually encountered when the other side is asymmetric nat.
>>
>> On Thu, Jun 23, 2016 at 11:20 AM, Agust? Ubalde <aubalde at presenceco.com
>> <javascript:_e(%7B%7D,'cvml','aubalde at presenceco.com');>> wrote:
>>
>>> HI all,
>>>
>>> In the log file I observed the trace *Changing Auto stun / rtp / DTLS*
>>> It only happens on connections from a particular IP. Anyone know explain to
>>> that is?
>>>
>>>
>>> Regards,
>>>
>>> *PRESENCE TECHNOLOGY*
>>> *Agust? Ubalde Bellot*
>>> Chief Developer
>>> C/ Comte Urgell 240 3A
>>> Barcelona 08036
>>> aubalde at presenceco.com
>>> <javascript:_e(%7B%7D,'cvml','aubalde at presenceco.com');>
>>>
>>> Ph: +34 93 10 10 300
>>> Fx: +34 93 10 10 333
>>>
>>> *www.presenceco.com* <http://www.presenceco.com/>
>>>
>>> *Follow us on:*
>>>
>>> *[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
>>> <http://www.youtube.com/PresenceTechnology>*[image: in]*
>>> <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]*
>>> <http://www.slideshare.net/PresenceTech>*[image: fb]*
>>> <http://www.facebook.com/presencetech>
>>>
>>> For additional information, please visit our website
>>> *www.presenceco.com* <http://www.presenceco.com/>
>>>
>>>
>>> *Presence Technology - DisclaimerThis message, its content and any file
>>> attached thereto is for the intended recipient only and is confidential and
>>> /or privileged. If you have received this e-mail in error or had access to
>>> it, you should note that the information in it is private and any use
>>> thereof is unauthorized. In such an event please notify us by e-mail or by
>>> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by
>>> whatsoever means and any transmission or dissemination thereof to other
>>> persons is prohibited. It should be deleted immediately from your system.
>>> Presence Technology reserves the right to take legal action against any
>>> persons unlawfully gaining access to the content of any external message it
>>> has emitted.*
>>>
>>> *For additional information, please visit our website **www.presenceco.com
>>> <http://www.presenceco.com/>*
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>>
>> ? http://freeswitch.org/  ? http://cluecon.com/  ?
>> http://twitter.com/FreeSWITCH
>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
>> <http://freeswitch.org/g+>*
>>
>> ClueCon Weekly Development Call
>> ? sip:888 at conference.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','sip:888 at conference.freeswitch.org');>  ?
>> +19193869900
>>
>> https://www.youtube.com/watch?v=9XXgW34t40s
>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> *Presence Technology - DisclaimerThis message, its content and any file
> attached thereto is for the intended recipient only and is confidential and
> /or privileged. If you have received this e-mail in error or had access to
> it, you should note that the information in it is private and any use
> thereof is unauthorized. In such an event please notify us by e-mail or by
> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by
> whatsoever means and any transmission or dissemination thereof to other
> persons is prohibited. It should be deleted immediately from your system.
> Presence Technology reserves the right to take legal action against any
> persons unlawfully gaining access to the content of any external message it
> has emitted.*
>
> *For additional information, please visit our website **www.presenceco.com
> <http://www.presenceco.com/>*
>
>
>


-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From anthony.minessale at gmail.com  Sun Jun 26 19:07:04 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 26 Jun 2016 10:07:04 -0500
Subject: [Freeswitch-users] Verto + external database
In-Reply-To: <CAEwTsAiSsqE4ZxYoZL+hX5gtUeK0zgmMk_4KF9YAC9Hk141LVA@mail.gmail.com>
References: <CAC7ttFEnV1iDf+7qbA9DJgfa+tcAotS9aN4fKRXB9gnYPcsFxg@mail.gmail.com>
	<CAEwTsAiSsqE4ZxYoZL+hX5gtUeK0zgmMk_4KF9YAC9Hk141LVA@mail.gmail.com>
Message-ID: <CAKbxfG-KU31_pZAvPq4cr8-CfGEkNMD2JXahQ_bLPAc3_SO=Zg@mail.gmail.com>

It might use the central registration db. I cant remember, try "show
registrations"

On Sunday, June 26, 2016, Michael Jerris <mike at jerris.com> wrote:

> verto stores registrations in memory,
> not in db.
>
> On Saturday, June 25, 2016, Gregor Nanger <gregor at infomedia.si
> <javascript:_e(%7B%7D,'cvml','gregor at infomedia.si');>> wrote:
>
>> I guess that verto uses db for storing user registrations.
>>
>> I am using external database (MS SQL) instead integrated sqlite, but I
>> cannot se verto users in registrations table.
>>
>> Does verto uses external db?
>>
>

-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From mike at jerris.com  Sun Jun 26 19:17:51 2016
From: mike at jerris.com (Michael Jerris)
Date: Sun, 26 Jun 2016 11:17:51 -0400
Subject: [Freeswitch-users] Changing Auto stun / rtp / DTLS
In-Reply-To: <CAKThY07KzmOo1f0CJiY6HeXQeQFmztRutZMhicb66C+tiuTjMw@mail.gmail.com>
References: <CAKThY069UjJHRuu9OL0Pi8CMhP=qR8V97RUirEWKwGWFC_8NkA@mail.gmail.com>
	<CAKbxfG8-6W-hT=cEdn=3QBCic8Os7FY1vadVA-r7m4r-6+yo5A@mail.gmail.com>
	<CAKThY07KzmOo1f0CJiY6HeXQeQFmztRutZMhicb66C+tiuTjMw@mail.gmail.com>
Message-ID: <CAEwTsAg+VX_6B=xdV1fQjcZrJPbG17ZCTaipj16FoK-jC2w9Ug@mail.gmail.com>

disruptions in network traffic might explain both of these.  You would have
to look at a packet trace to better understand what is going on completely.


On Sunday, June 26, 2016, Agust? Ubalde <aubalde at presenceco.com> wrote:

> Hi Anthony,
>
> This could cause that the connection works intermittently? Any idea to
> solve the problem?
>
> 2016-06-23 19:16 GMT+02:00 Anthony Minessale <anthony.minessale at gmail.com
> <javascript:_e(%7B%7D,'cvml','anthony.minessale at gmail.com');>>:
>
>> It happens when a candidate has not gotten any stun from the expected
>> places but it gets some from another place that is not the right address
>> but does have the right credentials.
>> It will lock on to the remote destination for that candidate instead.
>> This is usually encountered when the other side is asymmetric nat.
>>
>> On Thu, Jun 23, 2016 at 11:20 AM, Agust? Ubalde <aubalde at presenceco.com
>> <javascript:_e(%7B%7D,'cvml','aubalde at presenceco.com');>> wrote:
>>
>>> HI all,
>>>
>>> In the log file I observed the trace *Changing Auto stun / rtp / DTLS*
>>> It only happens on connections from a particular IP. Anyone know explain to
>>> that is?
>>>
>>
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From hunterj91 at hotmail.com  Sun Jun 26 19:54:07 2016
From: hunterj91 at hotmail.com (Jonathan Hunter)
Date: Sun, 26 Jun 2016 15:54:07 +0000
Subject: [Freeswitch-users] group_call>user_call>dial-string - Capture UUID ?
Message-ID: <DUB112-W13A6BCF4BB301E470902C8BD200@phx.gbl>

Hi All,
I am using a lua script to route to a hunt group of extensions using the group_call function, which then in turn actions a user_call, using dial-string.
After the dial-string has completed and FS creates an outgoing call such as;
switch_channel.c:1091 New Channel sofia/internal/test1 at testdomain [a076f82b-1db8-4385-914c-05dabc280d93]
Is it possible to grab the UUID created so I can store it in a database?
Many thanks
Jon 		 	   		  
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From gregor at infomedia.si  Sun Jun 26 19:55:49 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Sun, 26 Jun 2016 15:55:49 +0000
Subject: [Freeswitch-users] Verto + external database
In-Reply-To: <CAKbxfG-KU31_pZAvPq4cr8-CfGEkNMD2JXahQ_bLPAc3_SO=Zg@mail.gmail.com>
References: <CAC7ttFEnV1iDf+7qbA9DJgfa+tcAotS9aN4fKRXB9gnYPcsFxg@mail.gmail.com>
	<CAEwTsAiSsqE4ZxYoZL+hX5gtUeK0zgmMk_4KF9YAC9Hk141LVA@mail.gmail.com>
	<CAKbxfG-KU31_pZAvPq4cr8-CfGEkNMD2JXahQ_bLPAc3_SO=Zg@mail.gmail.com>
Message-ID: <CAC7ttFHd+owMxD93_AJGzMK_thG2oxkgBHg+Dxb1iy3zgk_gCQ@mail.gmail.com>

Ok, I get it.

Thank you for an answer.

On Sun, Jun 26, 2016, 17:07 Anthony Minessale <anthony.minessale at gmail.com>
wrote:

> It might use the central registration db. I cant remember, try "show
> registrations"
>
> On Sunday, June 26, 2016, Michael Jerris <mike at jerris.com> wrote:
>
>> verto stores registrations in memory,
>> not in db.
>>
>> On Saturday, June 25, 2016, Gregor Nanger <gregor at infomedia.si> wrote:
>>
>>> I guess that verto uses db for storing user registrations.
>>>
>>> I am using external database (MS SQL) instead integrated sqlite, but I
>>> cannot se verto users in registrations table.
>>>
>>> Does verto uses external db?
>>>
>>
>
> --
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
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
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From aubalde at presenceco.com  Mon Jun 27 00:25:07 2016
From: aubalde at presenceco.com (=?UTF-8?Q?Agust=C3=AD_Ubalde?=)
Date: Sun, 26 Jun 2016 22:25:07 +0200
Subject: [Freeswitch-users] Changing Auto stun / rtp / DTLS
In-Reply-To: <CAKbxfG__kyzdSc602vFCGUU7aS9NRPU7RfwpHKM0HzJ1GvxXaw@mail.gmail.com>
References: <CAKThY069UjJHRuu9OL0Pi8CMhP=qR8V97RUirEWKwGWFC_8NkA@mail.gmail.com>
	<CAKbxfG8-6W-hT=cEdn=3QBCic8Os7FY1vadVA-r7m4r-6+yo5A@mail.gmail.com>
	<CAKThY07KzmOo1f0CJiY6HeXQeQFmztRutZMhicb66C+tiuTjMw@mail.gmail.com>
	<CAKbxfG__kyzdSc602vFCGUU7aS9NRPU7RfwpHKM0HzJ1GvxXaw@mail.gmail.com>
Message-ID: <CAKThY05oERz+x9DSi86n7MfLneVtKxQdZkjdb-daGpteWwHy6Q@mail.gmail.com>

Hi Anthony,

The main problem is that there are connections suddenly stop receiving
audio.
It may be due to this scenario?


Thanks,

*PRESENCE TECHNOLOGY*
*Agust? Ubalde Bellot*
Chief Developer
C/ Comte Urgell 240 3A
Barcelona 08036
aubalde at presenceco.com

Ph: +34 93 10 10 300
Fx: +34 93 10 10 333

*www.presenceco.com* <http://www.presenceco.com/>

*Follow us on:*

*[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
<http://www.youtube.com/PresenceTechnology>*[image: in]*
<http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]*
<http://www.slideshare.net/PresenceTech>*[image: fb]*
<http://www.facebook.com/presencetech>

For additional information, please visit our website *www.presenceco.com*
<http://www.presenceco.com/>

2016-06-26 17:05 GMT+02:00 Anthony Minessale <anthony.minessale at gmail.com>:

> You didnt describe any problem really other than one host prints that
> message.  The bes solution is to enable symetric nat for that host.
>
>
>
> On Sunday, June 26, 2016, Agust? Ubalde <aubalde at presenceco.com> wrote:
>
>> Hi Anthony,
>>
>> This could cause that the connection works intermittently? Any idea to
>> solve the problem?
>>
>>
>> Thanks!
>>
>> *PRESENCE TECHNOLOGY*
>> *Agust? Ubalde Bellot*
>> Chief Developer
>> C/ Comte Urgell 240 3A
>> Barcelona 08036
>> aubalde at presenceco.com
>>
>> Ph: +34 93 10 10 300
>> Fx: +34 93 10 10 333
>>
>> *www.presenceco.com* <http://www.presenceco.com/>
>>
>> *Follow us on:*
>>
>> *[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
>> <http://www.youtube.com/PresenceTechnology>*[image: in]*
>> <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]*
>> <http://www.slideshare.net/PresenceTech>*[image: fb]*
>> <http://www.facebook.com/presencetech>
>>
>> For additional information, please visit our website *www.presenceco.com*
>> <http://www.presenceco.com/>
>>
>> 2016-06-23 19:16 GMT+02:00 Anthony Minessale <anthony.minessale at gmail.com
>> >:
>>
>>> It happens when a candidate has not gotten any stun from the expected
>>> places but it gets some from another place that is not the right address
>>> but does have the right credentials.
>>> It will lock on to the remote destination for that candidate instead.
>>> This is usually encountered when the other side is asymmetric nat.
>>>
>>> On Thu, Jun 23, 2016 at 11:20 AM, Agust? Ubalde <aubalde at presenceco.com>
>>> wrote:
>>>
>>>> HI all,
>>>>
>>>> In the log file I observed the trace *Changing Auto stun / rtp / DTLS*
>>>> It only happens on connections from a particular IP. Anyone know explain to
>>>> that is?
>>>>
>>>>
>>>> Regards,
>>>>
>>>> *PRESENCE TECHNOLOGY*
>>>> *Agust? Ubalde Bellot*
>>>> Chief Developer
>>>> C/ Comte Urgell 240 3A
>>>> Barcelona 08036
>>>> aubalde at presenceco.com
>>>>
>>>> Ph: +34 93 10 10 300
>>>> Fx: +34 93 10 10 333
>>>>
>>>> *www.presenceco.com* <http://www.presenceco.com/>
>>>>
>>>> *Follow us on:*
>>>>
>>>> *[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
>>>> <http://www.youtube.com/PresenceTechnology>*[image: in]*
>>>> <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]*
>>>> <http://www.slideshare.net/PresenceTech>*[image: fb]*
>>>> <http://www.facebook.com/presencetech>
>>>>
>>>> For additional information, please visit our website
>>>> *www.presenceco.com* <http://www.presenceco.com/>
>>>>
>>>>
>>>> *Presence Technology - DisclaimerThis message, its content and any file
>>>> attached thereto is for the intended recipient only and is confidential and
>>>> /or privileged. If you have received this e-mail in error or had access to
>>>> it, you should note that the information in it is private and any use
>>>> thereof is unauthorized. In such an event please notify us by e-mail or by
>>>> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by
>>>> whatsoever means and any transmission or dissemination thereof to other
>>>> persons is prohibited. It should be deleted immediately from your system.
>>>> Presence Technology reserves the right to take legal action against any
>>>> persons unlawfully gaining access to the content of any external message it
>>>> has emitted.*
>>>>
>>>> *For additional information, please visit our website **www.presenceco.com
>>>> <http://www.presenceco.com/>*
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>>>
>>> ? http://freeswitch.org/  ? http://cluecon.com/  ?
>>> http://twitter.com/FreeSWITCH
>>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
>>> <http://freeswitch.org/g+>*
>>>
>>> ClueCon Weekly Development Call
>>> ? sip:888 at conference.freeswitch.org  ? +19193869900
>>>
>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> *Presence Technology - DisclaimerThis message, its content and any file
>> attached thereto is for the intended recipient only and is confidential and
>> /or privileged. If you have received this e-mail in error or had access to
>> it, you should note that the information in it is private and any use
>> thereof is unauthorized. In such an event please notify us by e-mail or by
>> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by
>> whatsoever means and any transmission or dissemination thereof to other
>> persons is prohibited. It should be deleted immediately from your system.
>> Presence Technology reserves the right to take legal action against any
>> persons unlawfully gaining access to the content of any external message it
>> has emitted.*
>>
>> *For additional information, please visit our website **www.presenceco.com
>> <http://www.presenceco.com/>*
>>
>>
>>
>
>
> --
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 



*Presence Technology - DisclaimerThis message, its content and any file 
attached thereto is for the intended recipient only and is confidential and 
/or privileged. If you have received this e-mail in error or had access to 
it, you should note that the information in it is private and any use 
thereof is unauthorized. In such an event please notify us by e-mail or by 
telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by 
whatsoever means and any transmission or dissemination thereof to other 
persons is prohibited. It should be deleted immediately from your system. 
Presence Technology reserves the right to take legal action against any 
persons unlawfully gaining access to the content of any external message it 
has emitted.*

*For additional information, please visit our website **www.presenceco.com 
<http://www.presenceco.com/>*

 
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From mike at jerris.com  Mon Jun 27 01:10:04 2016
From: mike at jerris.com (Michael Jerris)
Date: Sun, 26 Jun 2016 17:10:04 -0400
Subject: [Freeswitch-users] Changing Auto stun / rtp / DTLS
In-Reply-To: <CAKThY05oERz+x9DSi86n7MfLneVtKxQdZkjdb-daGpteWwHy6Q@mail.gmail.com>
References: <CAKThY069UjJHRuu9OL0Pi8CMhP=qR8V97RUirEWKwGWFC_8NkA@mail.gmail.com>
	<CAKbxfG8-6W-hT=cEdn=3QBCic8Os7FY1vadVA-r7m4r-6+yo5A@mail.gmail.com>
	<CAKThY07KzmOo1f0CJiY6HeXQeQFmztRutZMhicb66C+tiuTjMw@mail.gmail.com>
	<CAKbxfG__kyzdSc602vFCGUU7aS9NRPU7RfwpHKM0HzJ1GvxXaw@mail.gmail.com>
	<CAKThY05oERz+x9DSi86n7MfLneVtKxQdZkjdb-daGpteWwHy6Q@mail.gmail.com>
Message-ID: <CAEwTsAj8Nh4y02isT6QAy17SBN_A_3mC0dGsfZJL96uqHArd-w@mail.gmail.com>

take a look at the packets and what happens when you stop receiving audio.

On Sunday, June 26, 2016, Agust? Ubalde <aubalde at presenceco.com> wrote:

> Hi Anthony,
>
> The main problem is that there are connections suddenly stop receiving
> audio.
> It may be due to this scenario?
>
>
> Thanks,
>
> *PRESENCE TECHNOLOGY*
> *Agust? Ubalde Bellot*
> Chief Developer
> C/ Comte Urgell 240 3A
> Barcelona 08036
> aubalde at presenceco.com
> <javascript:_e(%7B%7D,'cvml','aubalde at presenceco.com');>
>
> Ph: +34 93 10 10 300
> Fx: +34 93 10 10 333
>
> *www.presenceco.com* <http://www.presenceco.com/>
>
> *Follow us on:*
>
> *[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
> <http://www.youtube.com/PresenceTechnology>*[image: in]*
> <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]*
> <http://www.slideshare.net/PresenceTech>*[image: fb]*
> <http://www.facebook.com/presencetech>
>
> For additional information, please visit our website *www.presenceco.com*
> <http://www.presenceco.com/>
>
> 2016-06-26 17:05 GMT+02:00 Anthony Minessale <anthony.minessale at gmail.com
> <javascript:_e(%7B%7D,'cvml','anthony.minessale at gmail.com');>>:
>
>> You didnt describe any problem really other than one host prints that
>> message.  The bes solution is to enable symetric nat for that host.
>>
>>
>>
>> On Sunday, June 26, 2016, Agust? Ubalde <aubalde at presenceco.com
>> <javascript:_e(%7B%7D,'cvml','aubalde at presenceco.com');>> wrote:
>>
>>> Hi Anthony,
>>>
>>> This could cause that the connection works intermittently? Any idea to
>>> solve the problem?
>>>
>>>
>>> Thanks!
>>>
>>> *PRESENCE TECHNOLOGY*
>>> *Agust? Ubalde Bellot*
>>> Chief Developer
>>> C/ Comte Urgell 240 3A
>>> Barcelona 08036
>>> aubalde at presenceco.com
>>>
>>> Ph: +34 93 10 10 300
>>> Fx: +34 93 10 10 333
>>>
>>> *www.presenceco.com* <http://www.presenceco.com/>
>>>
>>> *Follow us on:*
>>>
>>> *[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
>>> <http://www.youtube.com/PresenceTechnology>*[image: in]*
>>> <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]*
>>> <http://www.slideshare.net/PresenceTech>*[image: fb]*
>>> <http://www.facebook.com/presencetech>
>>>
>>> For additional information, please visit our website
>>> *www.presenceco.com* <http://www.presenceco.com/>
>>>
>>> 2016-06-23 19:16 GMT+02:00 Anthony Minessale <
>>> anthony.minessale at gmail.com>:
>>>
>>>> It happens when a candidate has not gotten any stun from the expected
>>>> places but it gets some from another place that is not the right address
>>>> but does have the right credentials.
>>>> It will lock on to the remote destination for that candidate instead.
>>>> This is usually encountered when the other side is asymmetric nat.
>>>>
>>>> On Thu, Jun 23, 2016 at 11:20 AM, Agust? Ubalde <aubalde at presenceco.com
>>>> > wrote:
>>>>
>>>>> HI all,
>>>>>
>>>>> In the log file I observed the trace *Changing Auto stun / rtp / DTLS*
>>>>> It only happens on connections from a particular IP. Anyone know explain to
>>>>> that is?
>>>>>
>>>>>
>>>>> Regards,
>>>>>
>>>>> *PRESENCE TECHNOLOGY*
>>>>> *Agust? Ubalde Bellot*
>>>>> Chief Developer
>>>>> C/ Comte Urgell 240 3A
>>>>> Barcelona 08036
>>>>> aubalde at presenceco.com
>>>>>
>>>>> Ph: +34 93 10 10 300
>>>>> Fx: +34 93 10 10 333
>>>>>
>>>>> *www.presenceco.com* <http://www.presenceco.com/>
>>>>>
>>>>> *Follow us on:*
>>>>>
>>>>> *[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
>>>>> <http://www.youtube.com/PresenceTechnology>*[image: in]*
>>>>> <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]*
>>>>> <http://www.slideshare.net/PresenceTech>*[image: fb]*
>>>>> <http://www.facebook.com/presencetech>
>>>>>
>>>>> For additional information, please visit our website
>>>>> *www.presenceco.com* <http://www.presenceco.com/>
>>>>>
>>>>>
>>>>> *Presence Technology - DisclaimerThis message, its content and any
>>>>> file attached thereto is for the intended recipient only and is
>>>>> confidential and /or privileged. If you have received this e-mail in error
>>>>> or had access to it, you should note that the information in it is private
>>>>> and any use thereof is unauthorized. In such an event please notify us by
>>>>> e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail
>>>>> by whatsoever means and any transmission or dissemination thereof to other
>>>>> persons is prohibited. It should be deleted immediately from your system.
>>>>> Presence Technology reserves the right to take legal action against any
>>>>> persons unlawfully gaining access to the content of any external message it
>>>>> has emitted.*
>>>>>
>>>>> *For additional information, please visit our website **www.presenceco.com
>>>>> <http://www.presenceco.com/>*
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>>>>
>>>> ? http://freeswitch.org/  ? http://cluecon.com/  ?
>>>> http://twitter.com/FreeSWITCH
>>>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
>>>> <http://freeswitch.org/g+>*
>>>>
>>>> ClueCon Weekly Development Call
>>>> ? sip:888 at conference.freeswitch.org  ? +19193869900
>>>>
>>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> *Presence Technology - DisclaimerThis message, its content and any file
>>> attached thereto is for the intended recipient only and is confidential and
>>> /or privileged. If you have received this e-mail in error or had access to
>>> it, you should note that the information in it is private and any use
>>> thereof is unauthorized. In such an event please notify us by e-mail or by
>>> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by
>>> whatsoever means and any transmission or dissemination thereof to other
>>> persons is prohibited. It should be deleted immediately from your system.
>>> Presence Technology reserves the right to take legal action against any
>>> persons unlawfully gaining access to the content of any external message it
>>> has emitted.*
>>>
>>> *For additional information, please visit our website **www.presenceco.com
>>> <http://www.presenceco.com/>*
>>>
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>>
>> ? http://freeswitch.org/  ? http://cluecon.com/  ?
>> http://twitter.com/FreeSWITCH
>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
>> <http://freeswitch.org/g+>*
>>
>> ClueCon Weekly Development Call
>> ? sip:888 at conference.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','sip:888 at conference.freeswitch.org');>  ?
>> +19193869900
>>
>> https://www.youtube.com/watch?v=9XXgW34t40s
>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> *Presence Technology - DisclaimerThis message, its content and any file
> attached thereto is for the intended recipient only and is confidential and
> /or privileged. If you have received this e-mail in error or had access to
> it, you should note that the information in it is private and any use
> thereof is unauthorized. In such an event please notify us by e-mail or by
> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by
> whatsoever means and any transmission or dissemination thereof to other
> persons is prohibited. It should be deleted immediately from your system.
> Presence Technology reserves the right to take legal action against any
> persons unlawfully gaining access to the content of any external message it
> has emitted.*
>
> *For additional information, please visit our website **www.presenceco.com
> <http://www.presenceco.com/>*
>
>
>
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From happy.neko at gmail.com  Mon Jun 27 04:46:13 2016
From: happy.neko at gmail.com (happy.neko)
Date: Mon, 27 Jun 2016 03:46:13 +0300
Subject: [Freeswitch-users] Intercept command does not work in bypass media
	mode
Message-ID: <CAH=YQ8sLg4YvYF1b=MdS3DW=4i3pnRHkKwrJsZn2Y1os2CMcBw@mail.gmail.com>

Hi,

I need to implement this scenario:
inbound SIP call A is waiting indefinitely for inbound call B and then they
are bridged together. RTP media should flow directly from A to B and vice
versa while FreeSwitch should control SIP signaling.

I made some progress but I could not make media bypass mode to work.

My SIP profile settings:
    <param name="inbound-late-negotiation" value="true" />
    <param name="inbound-bypass-media" value="true" />

Dialplan for leg A:
    <!-- Somehow save leg A UUID.-->
    <action application="park" />

LUA script for leg B:
    leg_a_uuid = ...
    session:execute("intercept", leg_a_uuid)

This works but FreeSwitch is still in RTP media path with following error
in the log:
[ERR] switch_ivr.c:1718 Can't re-establsh media on sofia/myprofile/xyz
It worth mentioning that I do not answer or send early media before
executing intercept command so there is no prior media on both channel A
and B.

Any ideas what needs to be done to make this scenario work without RTP
proxying?

-- 
Best regards,
I. Sokolov
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From krice at freeswitch.org  Mon Jun 27 05:13:38 2016
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 26 Jun 2016 20:13:38 -0500
Subject: [Freeswitch-users] Intercept command does not work in bypass
	media mode
In-Reply-To: <CAH=YQ8sLg4YvYF1b=MdS3DW=4i3pnRHkKwrJsZn2Y1os2CMcBw@mail.gmail.com>
References: <CAH=YQ8sLg4YvYF1b=MdS3DW=4i3pnRHkKwrJsZn2Y1os2CMcBw@mail.gmail.com>
Message-ID: <EB915A41-207D-4AC4-813C-F7902101EB34@freeswitch.org>

Bypass media is a special mode, not all commands work with it. Pretty much any command that works with media does not work with it. You need to bring the media back to fs first

Sent from my iPhone

> On Jun 26, 2016, at 7:46 PM, happy.neko <happy.neko at gmail.com> wrote:
> 
> Hi,
> 
> I need to implement this scenario: 
> inbound SIP call A is waiting indefinitely for inbound call B and then they are bridged together. RTP media should flow directly from A to B and vice versa while FreeSwitch should control SIP signaling.
> 
> I made some progress but I could not make media bypass mode to work.
> 
> My SIP profile settings:
>     <param name="inbound-late-negotiation" value="true" />
>     <param name="inbound-bypass-media" value="true" />
> 
> Dialplan for leg A:
>     <!-- Somehow save leg A UUID.-->
>     <action application="park" />
> 
> LUA script for leg B:
>     leg_a_uuid = ...
>     session:execute("intercept", leg_a_uuid)
> 
> This works but FreeSwitch is still in RTP media path with following error in the log:
> [ERR] switch_ivr.c:1718 Can't re-establsh media on sofia/myprofile/xyz
> It worth mentioning that I do not answer or send early media before executing intercept command so there is no prior media on both channel A and B.
> 
> Any ideas what needs to be done to make this scenario work without RTP proxying?
> 
> -- 
> Best regards,
> I. Sokolov
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From k.presler at megafit.su  Mon Jun 27 05:20:14 2016
From: k.presler at megafit.su (k.presler at megafit.su)
Date: Mon, 27 Jun 2016 07:20:14 +0600
Subject: [Freeswitch-users] Wrong hangup cause in CDR
Message-ID: <7710141466990414@web24m.yandex.ru>

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From krice at freeswitch.org  Mon Jun 27 06:46:30 2016
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 26 Jun 2016 21:46:30 -0500
Subject: [Freeswitch-users] Wrong hangup cause in CDR
In-Reply-To: <7710141466990414@web24m.yandex.ru>
References: <7710141466990414@web24m.yandex.ru>
Message-ID: <70A0D167-FF55-4916-95B2-AA9A867D5D58@freeswitch.org>

Bugs get reported at https://freeswitch.org/jira

Sent from my iPhone

> On Jun 26, 2016, at 8:20 PM, k.presler at megafit.su wrote:
> 
> TSP responds with 404 (1, UNALLOCATED_NUMBER), so FS responds 404 to the client, and I see UNALLOCATED_NUMBER in the logs. Bun in CDR cause is 16 - NORMAL clearing, not 1, as it should.
>  
>  
>  
>  
> -- 
> ? ?????????, ??????? ?????? ?????????
> ??????? ??? ?????????
> e-mail: 112 at megafit.su
> ???: +7 (383) 311-09-09 (???.112)
>  
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From happy.neko at gmail.com  Mon Jun 27 07:32:30 2016
From: happy.neko at gmail.com (happy.neko)
Date: Mon, 27 Jun 2016 06:32:30 +0300
Subject: [Freeswitch-users] Intercept command does not work in bypass
 media mode
In-Reply-To: <EB915A41-207D-4AC4-813C-F7902101EB34@freeswitch.org>
References: <CAH=YQ8sLg4YvYF1b=MdS3DW=4i3pnRHkKwrJsZn2Y1os2CMcBw@mail.gmail.com>
	<EB915A41-207D-4AC4-813C-F7902101EB34@freeswitch.org>
Message-ID: <CAH=YQ8to2=0GVuM+8Rojj90ROx9PU2gppeNvn-UfrXrvCVvepA@mail.gmail.com>

2016-06-27 4:13 GMT+03:00 Ken Rice <krice at freeswitch.org>:

> Bypass media is a special mode, not all commands work with it. Pretty much
> any command that works with media does not work with it.
>

Does the same stands for Proxy Media mode ("inbound-proxy-media=true")?

You need to bring the media back to fs first
>

Please correct me if I am wrong: I should answer both legs first, than
bridge them with "intercept" command and move FreeSwitch out of the media
path with "uuid_media off uuid"?
However I thinks intercept is blocking command, how can I queue "uuid_media
of" from LUA script to be executed right after the call is intercepted?


> On Jun 26, 2016, at 7:46 PM, happy.neko <happy.neko at gmail.com> wrote:
>
> Hi,
>
> I need to implement this scenario:
> inbound SIP call A is waiting indefinitely for inbound call B and then
> they are bridged together. RTP media should flow directly from A to B and
> vice versa while FreeSwitch should control SIP signaling.
>
> I made some progress but I could not make media bypass mode to work.
>
> My SIP profile settings:
>     <param name="inbound-late-negotiation" value="true" />
>     <param name="inbound-bypass-media" value="true" />
>
> Dialplan for leg A:
>     <!-- Somehow save leg A UUID.-->
>     <action application="park" />
>
> LUA script for leg B:
>     leg_a_uuid = ...
>     session:execute("intercept", leg_a_uuid)
>
> This works but FreeSwitch is still in RTP media path with following error
> in the log:
> [ERR] switch_ivr.c:1718 Can't re-establsh media on sofia/myprofile/xyz
> It worth mentioning that I do not answer or send early media before
> executing intercept command so there is no prior media on both channel A
> and B.
>
> Any ideas what needs to be done to make this scenario work without RTP
> proxying?
>
> --
> Best regards,
> I. Sokolov
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From s.safarov at gmail.com  Mon Jun 27 07:34:38 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Mon, 27 Jun 2016 03:34:38 +0000
Subject: [Freeswitch-users] Wrong hangup cause in CDR
In-Reply-To: <7710141466990414@web24m.yandex.ru>
References: <7710141466990414@web24m.yandex.ru>
Message-ID: <CAHtxdDeMPf6hb5G4LnN7fgieDYmwtJbXB9GL1mODg2F1wUqMEg@mail.gmail.com>

If loopback endpoint used in dialplan then looks at FS-9269
<https://freeswitch.org/jira/browse/FS-9269>

Sergey

??, 27 ???. 2016 ?. ? 4:21, <k.presler at megafit.su>:

> TSP responds with 404 (1, UNALLOCATED_NUMBER), so FS responds 404 to the
> client, and I see *UNALLOCATED_NUMBER* in the logs. Bun in CDR cause is
> 16 - NORMAL clearing, not 1, as it should.
>
>
>
>
> --
> ? ?????????, ??????? ?????? ?????????
> ??????? ??? ?????????
> e-mail: 112 at megafit.su
> ???: +7 (383) 311-09-09 (???.112)
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From avi at avimarcus.net  Mon Jun 27 08:16:13 2016
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 27 Jun 2016 04:16:13 +0000
Subject: [Freeswitch-users] How to: drop inband dtmf
Message-ID: <0100015590126914-c138950e-be4a-4d6d-ac5f-f56628180c86-000000@email.amazonses.com>

I'm not sure this is posted anywhere, so I wanted to make sure it's here.

I have a carrier that sends both inband and RFC2833 DTMF.
For FreeSWITCH, this is fine, but if I bridge it on to someone else, then
the remote system often processes both for duplicate DTMF.

Here's a local fix that works for me (but you can't be in bypass mode) to
remove the inband (it always had rfc2833, but not *always *inband.)
Set this for the affected leg:

<action application="set" data="rtp_manual_rtp_bugs=FLUSH_JB_ON_DTMF"/>
> <action application="set" data="jitterbuffer_msec=60"/>
> <action application="set" data="sip_jitter_buffer_during_bridge=true"/>


I have in my notes: <action application="export"
data="nolocal:drop_dtmf=true"/> but I don't see on the wiki that it's for
sure that it works on inband, a quick look at the code and I still wasn't
sure.

-Avi Marcus
BestFone
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From krice at freeswitch.org  Mon Jun 27 08:52:12 2016
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 26 Jun 2016 23:52:12 -0500
Subject: [Freeswitch-users] Intercept command does not work in bypass
	media mode
In-Reply-To: <CAH=YQ8to2=0GVuM+8Rojj90ROx9PU2gppeNvn-UfrXrvCVvepA@mail.gmail.com>
References: <CAH=YQ8sLg4YvYF1b=MdS3DW=4i3pnRHkKwrJsZn2Y1os2CMcBw@mail.gmail.com>
	<EB915A41-207D-4AC4-813C-F7902101EB34@freeswitch.org>
	<CAH=YQ8to2=0GVuM+8Rojj90ROx9PU2gppeNvn-UfrXrvCVvepA@mail.gmail.com>
Message-ID: <FC5A317D-D290-47CA-852D-CAB88AE60640@freeswitch.org>


See uuid_bridge note most of the time you shouldnt be doing things like this in scripts as it leaves an interpreter running per call

Proxy mode is not really worth it.. If you are just avoiding trancoding normal mode is just as efficient

Sent from my iPhone

> On Jun 26, 2016, at 10:32 PM, happy.neko <happy.neko at gmail.com> wrote:
> 
> 2016-06-27 4:13 GMT+03:00 Ken Rice <krice at freeswitch.org>:
>> Bypass media is a special mode, not all commands work with it. Pretty much any command that works with media does not work with it.
> 
> Does the same stands for Proxy Media mode ("inbound-proxy-media=true")?
> 
>> You need to bring the media back to fs first
> 
> Please correct me if I am wrong: I should answer both legs first, than bridge them with "intercept" command and move FreeSwitch out of the media path with "uuid_media off uuid"?
> However I thinks intercept is blocking command, how can I queue "uuid_media of" from LUA script to be executed right after the call is intercepted?
>  
>>> On Jun 26, 2016, at 7:46 PM, happy.neko <happy.neko at gmail.com> wrote:
>>> 
>>> Hi,
>>> 
>>> I need to implement this scenario: 
>>> inbound SIP call A is waiting indefinitely for inbound call B and then they are bridged together. RTP media should flow directly from A to B and vice versa while FreeSwitch should control SIP signaling.
>>> 
>>> I made some progress but I could not make media bypass mode to work.
>>> 
>>> My SIP profile settings:
>>>     <param name="inbound-late-negotiation" value="true" />
>>>     <param name="inbound-bypass-media" value="true" />
>>> 
>>> Dialplan for leg A:
>>>     <!-- Somehow save leg A UUID.-->
>>>     <action application="park" />
>>> 
>>> LUA script for leg B:
>>>     leg_a_uuid = ...
>>>     session:execute("intercept", leg_a_uuid)
>>> 
>>> This works but FreeSwitch is still in RTP media path with following error in the log:
>>> [ERR] switch_ivr.c:1718 Can't re-establsh media on sofia/myprofile/xyz
>>> It worth mentioning that I do not answer or send early media before executing intercept command so there is no prior media on both channel A and B.
>>> 
>>> Any ideas what needs to be done to make this scenario work without RTP proxying?
>>> 
>>> -- 
>>> Best regards,
>>> I. Sokolov
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services: 
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From happy.neko at gmail.com  Mon Jun 27 09:08:05 2016
From: happy.neko at gmail.com (happy.neko)
Date: Mon, 27 Jun 2016 08:08:05 +0300
Subject: [Freeswitch-users] Intercept command does not work in bypass
 media mode
In-Reply-To: <FC5A317D-D290-47CA-852D-CAB88AE60640@freeswitch.org>
References: <CAH=YQ8sLg4YvYF1b=MdS3DW=4i3pnRHkKwrJsZn2Y1os2CMcBw@mail.gmail.com>
	<EB915A41-207D-4AC4-813C-F7902101EB34@freeswitch.org>
	<CAH=YQ8to2=0GVuM+8Rojj90ROx9PU2gppeNvn-UfrXrvCVvepA@mail.gmail.com>
	<FC5A317D-D290-47CA-852D-CAB88AE60640@freeswitch.org>
Message-ID: <CAH=YQ8t5qPpvWj84qgXYK8YVRPLOe+WKa15eCFqszH_+mWV7_w@mail.gmail.com>

2016-06-27 7:52 GMT+03:00 Ken Rice <krice at freeswitch.org>:

>
> See uuid_bridge note most of the time you shouldnt be doing things like
> this in scripts as it leaves an interpreter running per call
>

Please elaborate more about interpreter thing. If I bridge calls with
intercept / uuid_bridge command in LUA script and have 1000 concurrent
calls I'll end up with 1000 idle threads?
What is your advice about doing it more efficiently: execute intercept
command from XML dialplan (does it use event loop?), external control
through event socket?

Proxy mode is not really worth it.. If you are just avoiding trancoding
> normal mode is just as efficient
>
> Sent from my iPhone
>
> On Jun 26, 2016, at 10:32 PM, happy.neko <happy.neko at gmail.com> wrote:
>
> 2016-06-27 4:13 GMT+03:00 Ken Rice <krice at freeswitch.org>:
>
>> Bypass media is a special mode, not all commands work with it. Pretty
>> much any command that works with media does not work with it.
>>
>
> Does the same stands for Proxy Media mode ("inbound-proxy-media=true")?
>
> You need to bring the media back to fs first
>>
>
> Please correct me if I am wrong: I should answer both legs first, than
> bridge them with "intercept" command and move FreeSwitch out of the media
> path with "uuid_media off uuid"?
> However I thinks intercept is blocking command, how can I queue
> "uuid_media of" from LUA script to be executed right after the call is
> intercepted?
>
>
>> On Jun 26, 2016, at 7:46 PM, happy.neko <happy.neko at gmail.com> wrote:
>>
>> Hi,
>>
>> I need to implement this scenario:
>> inbound SIP call A is waiting indefinitely for inbound call B and then
>> they are bridged together. RTP media should flow directly from A to B and
>> vice versa while FreeSwitch should control SIP signaling.
>>
>> I made some progress but I could not make media bypass mode to work.
>>
>> My SIP profile settings:
>>     <param name="inbound-late-negotiation" value="true" />
>>     <param name="inbound-bypass-media" value="true" />
>>
>> Dialplan for leg A:
>>     <!-- Somehow save leg A UUID.-->
>>     <action application="park" />
>>
>> LUA script for leg B:
>>     leg_a_uuid = ...
>>     session:execute("intercept", leg_a_uuid)
>>
>> This works but FreeSwitch is still in RTP media path with following error
>> in the log:
>> [ERR] switch_ivr.c:1718 Can't re-establsh media on sofia/myprofile/xyz
>> It worth mentioning that I do not answer or send early media before
>> executing intercept command so there is no prior media on both channel A
>> and B.
>>
>> Any ideas what needs to be done to make this scenario work without RTP
>> proxying?
>>
>> --
>> Best regards,
>> I. Sokolov
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gb at cm.nl  Mon Jun 27 11:49:12 2016
From: gb at cm.nl (Grant Bagdasarian)
Date: Mon, 27 Jun 2016 07:49:12 +0000
Subject: [Freeswitch-users] Disable INFO messages
Message-ID: <0ece12049f594983bbb3e7d62aea44c9@CM-EX-V01.cm.local>

Hello,

How do I disable INFO messages sent by Freeswitch?
I have no idea why  these messages are sent and I would like to disable them completely.

Regards,

Grant
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From christophe.yann at gmail.com  Mon Jun 27 13:21:44 2016
From: christophe.yann at gmail.com (yann christophe)
Date: Mon, 27 Jun 2016 11:21:44 +0200
Subject: [Freeswitch-users] Freeswitch on AWS
In-Reply-To: <CAC8JMOJFLXJHEFyEcSAB-2nY6pLuAKWziyJQTFFGukOun5j6bg@mail.gmail.com>
References: <CA+qsECtiQ6scQLc-htQngbqprfdavWdc7QQ3OLEC2Bujy0pXVg@mail.gmail.com>
	<CAC8JMOJFLXJHEFyEcSAB-2nY6pLuAKWziyJQTFFGukOun5j6bg@mail.gmail.com>
Message-ID: <CA+qsECuTBCg8Lsv4ayOHTVeFcVfPf7wjicZvsfVHJeR7UF2Nvw@mail.gmail.com>

Hello,

Thanks Oleg, if i undersand, i can copy the file from one FS server to
S3 storage and copy this file from S3 to other FS server ?

Thanks Tristan for the advise about CPU.

Best regards

2016-06-24 15:21 GMT+02:00 Oleg Stolyar <olegstolyar at gmail.com>:
> I regularly transfer large files between my FS servers in EC2 and S3 using
> AWS s3 command line utility
> (http://docs.aws.amazon.com/cli/latest/reference/s3/index.html).
>
> The sync operation called from a cron job is what you are probably looking
> for to replace rsync.
>
>
> On Fri, Jun 24, 2016 at 2:07 AM, yann christophe <christophe.yann at gmail.com>
> wrote:
>>
>> Hello everyone,
>>
>> I would like to know if someone has an experience on AWS with Freeswitch.
>>
>> Because i'll deploy 2 Freeswitch  servers on EC2 and i would like to
>> use S3 for the directory "Dialplan". I want this Directory is the same
>> for these 2 servers,  I currently use rsync, maybe it's better to use
>> S3 for that.
>>
>> What do you suggest me ? I just started to work on AWS
>>
>> Thanks
>> Best regards
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From gaz.foreman at gmail.com  Mon Jun 27 16:40:45 2016
From: gaz.foreman at gmail.com (Gary Foreman)
Date: Mon, 27 Jun 2016 13:40:45 +0100
Subject: [Freeswitch-users] No ringback / ESL channel progress when using
	originate
Message-ID: <CAOOB3LaT8hjnVufWh_BHukkffy_PM6X_sabZb83aMwbi=dav8A@mail.gmail.com>

Hi all,

Ever since upgrading past 1.6.6 I've had problems when using the originate
command.

I'm unable to hear ringback on the call and no channel_progress ESL event
is emitted when the "183 Session Progress" message comes back from my SIP
provider.

Once the call is connected I can talk as usual.

Here are the console outputs;

1.6.6 (working)

https://pastebin.freeswitch.org/view/3998aa81

1.6.9 (no ringback)

https://pastebin.freeswitch.org/view/65002d96

Thanks in advance
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From olegstolyar at gmail.com  Mon Jun 27 17:40:06 2016
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Mon, 27 Jun 2016 06:40:06 -0700
Subject: [Freeswitch-users] Freeswitch on AWS
In-Reply-To: <CA+qsECuTBCg8Lsv4ayOHTVeFcVfPf7wjicZvsfVHJeR7UF2Nvw@mail.gmail.com>
References: <CA+qsECtiQ6scQLc-htQngbqprfdavWdc7QQ3OLEC2Bujy0pXVg@mail.gmail.com>
	<CAC8JMOJFLXJHEFyEcSAB-2nY6pLuAKWziyJQTFFGukOun5j6bg@mail.gmail.com>
	<CA+qsECuTBCg8Lsv4ayOHTVeFcVfPf7wjicZvsfVHJeR7UF2Nvw@mail.gmail.com>
Message-ID: <CAC8JMOK-tybb-KLjq+ufnsbUFhbCB1rrEBQA223jj9EtBB-kEQ@mail.gmail.com>

Yes, you can.
On Jun 27, 2016 2:24 AM, "yann christophe" <christophe.yann at gmail.com>
wrote:

> Hello,
>
> Thanks Oleg, if i undersand, i can copy the file from one FS server to
> S3 storage and copy this file from S3 to other FS server ?
>
> Thanks Tristan for the advise about CPU.
>
> Best regards
>
> 2016-06-24 15:21 GMT+02:00 Oleg Stolyar <olegstolyar at gmail.com>:
> > I regularly transfer large files between my FS servers in EC2 and S3
> using
> > AWS s3 command line utility
> > (http://docs.aws.amazon.com/cli/latest/reference/s3/index.html).
> >
> > The sync operation called from a cron job is what you are probably
> looking
> > for to replace rsync.
> >
> >
> > On Fri, Jun 24, 2016 at 2:07 AM, yann christophe <
> christophe.yann at gmail.com>
> > wrote:
> >>
> >> Hello everyone,
> >>
> >> I would like to know if someone has an experience on AWS with
> Freeswitch.
> >>
> >> Because i'll deploy 2 Freeswitch  servers on EC2 and i would like to
> >> use S3 for the directory "Dialplan". I want this Directory is the same
> >> for these 2 servers,  I currently use rsync, maybe it's better to use
> >> S3 for that.
> >>
> >> What do you suggest me ? I just started to work on AWS
> >>
> >> Thanks
> >> Best regards
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon Jun 27 18:36:09 2016
From: brian at freeswitch.org (Brian West)
Date: Mon, 27 Jun 2016 09:36:09 -0500
Subject: [Freeswitch-users] Intercept command does not work in bypass
 media mode
In-Reply-To: <CAH=YQ8t5qPpvWj84qgXYK8YVRPLOe+WKa15eCFqszH_+mWV7_w@mail.gmail.com>
References: <CAH=YQ8sLg4YvYF1b=MdS3DW=4i3pnRHkKwrJsZn2Y1os2CMcBw@mail.gmail.com>
	<EB915A41-207D-4AC4-813C-F7902101EB34@freeswitch.org>
	<CAH=YQ8to2=0GVuM+8Rojj90ROx9PU2gppeNvn-UfrXrvCVvepA@mail.gmail.com>
	<FC5A317D-D290-47CA-852D-CAB88AE60640@freeswitch.org>
	<CAH=YQ8t5qPpvWj84qgXYK8YVRPLOe+WKa15eCFqszH_+mWV7_w@mail.gmail.com>
Message-ID: <CAEJMVkDt+RxE1DZ4MHw6q4qT3BO91hBODyTC4ZWgxLM3JqmnWA@mail.gmail.com>

If you're in proxy media mode, (WHICH YOU SHOULD NOT USE), then nothing can
touch the media, intercept, recording, various other media related commands
will not function as expected.

On Mon, Jun 27, 2016 at 12:08 AM, happy.neko <happy.neko at gmail.com> wrote:

> 2016-06-27 7:52 GMT+03:00 Ken Rice <krice at freeswitch.org>:
>
>>
>> See uuid_bridge note most of the time you shouldnt be doing things like
>> this in scripts as it leaves an interpreter running per call
>>
>
> Please elaborate more about interpreter thing. If I bridge calls with
> intercept / uuid_bridge command in LUA script and have 1000 concurrent
> calls I'll end up with 1000 idle threads?
> What is your advice about doing it more efficiently: execute intercept
> command from XML dialplan (does it use event loop?), external control
> through event socket?
>
> Proxy mode is not really worth it.. If you are just avoiding trancoding
>> normal mode is just as efficient
>>
>> Sent from my iPhone
>>
>> On Jun 26, 2016, at 10:32 PM, happy.neko <happy.neko at gmail.com> wrote:
>>
>> 2016-06-27 4:13 GMT+03:00 Ken Rice <krice at freeswitch.org>:
>>
>>> Bypass media is a special mode, not all commands work with it. Pretty
>>> much any command that works with media does not work with it.
>>>
>>
>> Does the same stands for Proxy Media mode ("inbound-proxy-media=true")?
>>
>> You need to bring the media back to fs first
>>>
>>
>> Please correct me if I am wrong: I should answer both legs first, than
>> bridge them with "intercept" command and move FreeSwitch out of the media
>> path with "uuid_media off uuid"?
>> However I thinks intercept is blocking command, how can I queue
>> "uuid_media of" from LUA script to be executed right after the call is
>> intercepted?
>>
>>
>>> On Jun 26, 2016, at 7:46 PM, happy.neko <happy.neko at gmail.com> wrote:
>>>
>>> Hi,
>>>
>>> I need to implement this scenario:
>>> inbound SIP call A is waiting indefinitely for inbound call B and then
>>> they are bridged together. RTP media should flow directly from A to B and
>>> vice versa while FreeSwitch should control SIP signaling.
>>>
>>> I made some progress but I could not make media bypass mode to work.
>>>
>>> My SIP profile settings:
>>>     <param name="inbound-late-negotiation" value="true" />
>>>     <param name="inbound-bypass-media" value="true" />
>>>
>>> Dialplan for leg A:
>>>     <!-- Somehow save leg A UUID.-->
>>>     <action application="park" />
>>>
>>> LUA script for leg B:
>>>     leg_a_uuid = ...
>>>     session:execute("intercept", leg_a_uuid)
>>>
>>> This works but FreeSwitch is still in RTP media path with following
>>> error in the log:
>>> [ERR] switch_ivr.c:1718 Can't re-establsh media on sofia/myprofile/xyz
>>> It worth mentioning that I do not answer or send early media before
>>> executing intercept command so there is no prior media on both channel A
>>> and B.
>>>
>>> Any ideas what needs to be done to make this scenario work without RTP
>>> proxying?
>>>
>>> --
>>> Best regards,
>>> I. Sokolov
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com
https://www.gofundme.com/freeswitch_ubuntu

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

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From giggsey at gmail.com  Mon Jun 27 19:40:55 2016
From: giggsey at gmail.com (Joshua Gigg)
Date: Mon, 27 Jun 2016 15:40:55 +0000
Subject: [Freeswitch-users] Disable INFO messages
In-Reply-To: <0ece12049f594983bbb3e7d62aea44c9@CM-EX-V01.cm.local>
References: <0ece12049f594983bbb3e7d62aea44c9@CM-EX-V01.cm.local>
Message-ID: <CADcg8iWYdvNhCieLo0s1JSEY6t85ZjOXPzyX3jc7p_NhZT8oig@mail.gmail.com>

Do you mean when the Caller ID updates (on bridge) etc?

If so, take a look at send-display-update (
https://wiki.freeswitch.org/wiki/Sofia.conf.xml#send-display-update)

On Mon, 27 Jun 2016 at 08:52 Grant Bagdasarian <gb at cm.nl> wrote:

> Hello,
>
>
>
> How do I disable INFO messages sent by Freeswitch?
>
> I have no idea why  these messages are sent and I would like to disable
> them completely.
>
>
>
> Regards,
>
>
>
> Grant
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mike at jerris.com  Mon Jun 27 20:07:26 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 27 Jun 2016 12:07:26 -0400
Subject: [Freeswitch-users] Intercept command does not work in bypass
 media mode
In-Reply-To: <CAEJMVkDt+RxE1DZ4MHw6q4qT3BO91hBODyTC4ZWgxLM3JqmnWA@mail.gmail.com>
References: <CAH=YQ8sLg4YvYF1b=MdS3DW=4i3pnRHkKwrJsZn2Y1os2CMcBw@mail.gmail.com>
	<EB915A41-207D-4AC4-813C-F7902101EB34@freeswitch.org>
	<CAH=YQ8to2=0GVuM+8Rojj90ROx9PU2gppeNvn-UfrXrvCVvepA@mail.gmail.com>
	<FC5A317D-D290-47CA-852D-CAB88AE60640@freeswitch.org>
	<CAH=YQ8t5qPpvWj84qgXYK8YVRPLOe+WKa15eCFqszH_+mWV7_w@mail.gmail.com>
	<CAEJMVkDt+RxE1DZ4MHw6q4qT3BO91hBODyTC4ZWgxLM3JqmnWA@mail.gmail.com>
Message-ID: <CAEwTsAiOiB407uArqVwbGRQ6AD8mUVW8bF4Z82NVjfnh6hyN-A@mail.gmail.com>

can't you use bypass media after bridge?

On Monday, June 27, 2016, Brian West <brian at freeswitch.org> wrote:

> If you're in proxy media mode, (WHICH YOU SHOULD NOT USE), then nothing
> can touch the media, intercept, recording, various other media related
> commands will not function as expected.
>
> On Mon, Jun 27, 2016 at 12:08 AM, happy.neko <happy.neko at gmail.com
> <javascript:_e(%7B%7D,'cvml','happy.neko at gmail.com');>> wrote:
>
>> 2016-06-27 7:52 GMT+03:00 Ken Rice <krice at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','krice at freeswitch.org');>>:
>>
>>>
>>> See uuid_bridge note most of the time you shouldnt be doing things like
>>> this in scripts as it leaves an interpreter running per call
>>>
>>
>> Please elaborate more about interpreter thing. If I bridge calls with
>> intercept / uuid_bridge command in LUA script and have 1000 concurrent
>> calls I'll end up with 1000 idle threads?
>> What is your advice about doing it more efficiently: execute intercept
>> command from XML dialplan (does it use event loop?), external control
>> through event socket?
>>
>> Proxy mode is not really worth it.. If you are just avoiding trancoding
>>> normal mode is just as efficient
>>>
>>> Sent from my iPhone
>>>
>>> On Jun 26, 2016, at 10:32 PM, happy.neko <happy.neko at gmail.com
>>> <javascript:_e(%7B%7D,'cvml','happy.neko at gmail.com');>> wrote:
>>>
>>> 2016-06-27 4:13 GMT+03:00 Ken Rice <krice at freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','krice at freeswitch.org');>>:
>>>
>>>> Bypass media is a special mode, not all commands work with it. Pretty
>>>> much any command that works with media does not work with it.
>>>>
>>>
>>> Does the same stands for Proxy Media mode ("inbound-proxy-media=true")?
>>>
>>> You need to bring the media back to fs first
>>>>
>>>
>>> Please correct me if I am wrong: I should answer both legs first, than
>>> bridge them with "intercept" command and move FreeSwitch out of the media
>>> path with "uuid_media off uuid"?
>>> However I thinks intercept is blocking command, how can I queue
>>> "uuid_media of" from LUA script to be executed right after the call is
>>> intercepted?
>>>
>>>
>>>> On Jun 26, 2016, at 7:46 PM, happy.neko <happy.neko at gmail.com
>>>> <javascript:_e(%7B%7D,'cvml','happy.neko at gmail.com');>> wrote:
>>>>
>>>> Hi,
>>>>
>>>> I need to implement this scenario:
>>>> inbound SIP call A is waiting indefinitely for inbound call B and then
>>>> they are bridged together. RTP media should flow directly from A to B and
>>>> vice versa while FreeSwitch should control SIP signaling.
>>>>
>>>> I made some progress but I could not make media bypass mode to work.
>>>>
>>>> My SIP profile settings:
>>>>     <param name="inbound-late-negotiation" value="true" />
>>>>     <param name="inbound-bypass-media" value="true" />
>>>>
>>>> Dialplan for leg A:
>>>>     <!-- Somehow save leg A UUID.-->
>>>>     <action application="park" />
>>>>
>>>> LUA script for leg B:
>>>>     leg_a_uuid = ...
>>>>     session:execute("intercept", leg_a_uuid)
>>>>
>>>> This works but FreeSwitch is still in RTP media path with following
>>>> error in the log:
>>>> [ERR] switch_ivr.c:1718 Can't re-establsh media on sofia/myprofile/xyz
>>>> It worth mentioning that I do not answer or send early media before
>>>> executing intercept command so there is no prior media on both channel A
>>>> and B.
>>>>
>>>> Any ideas what needs to be done to make this scenario work without RTP
>>>> proxying?
>>>>
>>>> --
>>>> Best regards,
>>>> I. Sokolov
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://
>>>> lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','brian at freeswitch.org');>
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
> https://www.gofundme.com/freeswitch_ubuntu
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
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From marketing at cluecon.com  Mon Jun 27 23:20:53 2016
From: marketing at cluecon.com (Kathleen King)
Date: Mon, 27 Jun 2016 12:20:53 -0700
Subject: [Freeswitch-users] Join us for the Official ClueCon After Party
	hosted by Telnyx!
Message-ID: <CAOStUPZeWrK9zE0RGSWbM-h5Zrkv-9=w7NADHmxB3+tSfGYBiA@mail.gmail.com>

[image: Banner] <#>

Come and enjoy drinks and apps on us, while networking with your peers and
enjoying live music, football, ping pong, and old school video games along
our 30? media wall.

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[image: When Icon]

When

Wednesday, August 10, 2016 from 7:00 PM ? 12:00 AM
[image: Where Icon]

Where

311 W Superior St, Ste 504
Chicago, IL 60654

Located less than 2 miles from Swissotel a temporary shuttle service will
be available.
[image: What Icon]

What
 Drinks, Food, Games, Networking and Special Giveaways!
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From gb at cm.nl  Tue Jun 28 15:40:19 2016
From: gb at cm.nl (Grant Bagdasarian)
Date: Tue, 28 Jun 2016 11:40:19 +0000
Subject: [Freeswitch-users] FS not relaying 180/183 in proxy_media
Message-ID: <ada0d1fd3f3c4ea389d28a40cb26bb9f@CM-EX-V01.cm.local>

Hello,

Is it possible to make freeswitch relay the 180 or 183 replies, including the ringback tone, in proxy_media mode to the next hop?

Regards,

Grant
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From john.nash778 at gmail.com  Tue Jun 28 18:16:54 2016
From: john.nash778 at gmail.com (John Nash)
Date: Tue, 28 Jun 2016 19:46:54 +0530
Subject: [Freeswitch-users] wss listener not starting
Message-ID: <CAA4rGPxVFE=949-Q8c=RbByWhishu6fqztL31tLx7xBLc1BtaA@mail.gmail.com>

I have been using freeswitch for udp and today I created a new profile and
uncommented wss line and configured as per
https://freeswitch.org/confluence/display/FREESWITCH/WebRTC

But after i rstarted and tried to check sofia status I do not see wss
listener (I see udp and TLS) In log also I do not see any error or anything
related to wss (INFO level)

Am I missing somthing obvious?...any module dependency for wss?
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From krice at freeswitch.org  Tue Jun 28 18:34:14 2016
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 28 Jun 2016 09:34:14 -0500
Subject: [Freeswitch-users] FS not relaying 180/183 in proxy_media
In-Reply-To: <ada0d1fd3f3c4ea389d28a40cb26bb9f@CM-EX-V01.cm.local>
References: <ada0d1fd3f3c4ea389d28a40cb26bb9f@CM-EX-V01.cm.local>
Message-ID: <89d501d1d14a$25a795e0$70f6c1a0$@freeswitch.org>

FS auto-relays those in any mode. however proxy media itself has no real
advantage over just setting late negotiation which reduces or eliminates
transcoding. Transcoding is where most of the load is there. as far as the
180/183 as long as its 1:1 call in to call out from the FS perspective those
are passed across the bridge automatically. if you are doing something like
get a call in then originate another call and trying to manually bridge them
after the b leg is answered, this will not work and is recommended against
generally.

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Grant
Bagdasarian
Sent: Tuesday, June 28, 2016 6:40 AM
To: FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org)
<freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] FS not relaying 180/183 in proxy_media

 

Hello,

 

Is it possible to make freeswitch relay the 180 or 183 replies, including
the ringback tone, in proxy_media mode to the next hop?

 

Regards,

 

Grant 

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From mirkobrankovic at gmail.com  Tue Jun 28 18:37:56 2016
From: mirkobrankovic at gmail.com (Mirko Brankovic)
Date: Tue, 28 Jun 2016 16:37:56 +0200
Subject: [Freeswitch-users] wss listener not starting
In-Reply-To: <CAA4rGPxVFE=949-Q8c=RbByWhishu6fqztL31tLx7xBLc1BtaA@mail.gmail.com>
References: <CAA4rGPxVFE=949-Q8c=RbByWhishu6fqztL31tLx7xBLc1BtaA@mail.gmail.com>
Message-ID: <CAND18T21EhRANNkfKaVwQZODsByf3JmtChZ8iW1p3bp8aXV+3A@mail.gmail.com>

Hi John,

I guess all you need is:
    <param name="wss-binding" value=":<your_port>"/>
and proper wss.pem in certs/ dir

What does  WSS-BIND-URL says on sofia status profile <your_new_profile>, if
it starts at all ?


On Tue, Jun 28, 2016 at 4:16 PM, John Nash <john.nash778 at gmail.com> wrote:

> I have been using freeswitch for udp and today I created a new profile and
> uncommented wss line and configured as per
> https://freeswitch.org/confluence/display/FREESWITCH/WebRTC
>
> But after i rstarted and tried to check sofia status I do not see wss
> listener (I see udp and TLS) In log also I do not see any error or anything
> related to wss (INFO level)
>
> Am I missing somthing obvious?...any module dependency for wss?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Regards,
Mirko
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From s.safarov at gmail.com  Tue Jun 28 18:41:41 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 28 Jun 2016 14:41:41 +0000
Subject: [Freeswitch-users] FS not relaying 180/183 in proxy_media
In-Reply-To: <89d501d1d14a$25a795e0$70f6c1a0$@freeswitch.org>
References: <ada0d1fd3f3c4ea389d28a40cb26bb9f@CM-EX-V01.cm.local>
	<89d501d1d14a$25a795e0$70f6c1a0$@freeswitch.org>
Message-ID: <CAHtxdDc4x3gbsW6eGRzWRbeYUxHonwxWBb1WFDiWAH4P+TaipQ@mail.gmail.com>

Current FS requires proxy media if you use zrtp and exist NAT in media path.

On Tue, Jun 28, 2016, 17:35 Ken Rice <krice at freeswitch.org> wrote:

> FS auto-relays those in any mode? however proxy media itself has no real
> advantage over just setting late negotiation which reduces or eliminates
> transcoding. Transcoding is where most of the load is there? as far as the
> 180/183 as long as its 1:1 call in to call out from the FS perspective
> those are passed across the bridge automatically? if you are doing
> something like get a call in then originate another call and trying to
> manually bridge them after the b leg is answered, this will not work and is
> recommended against generally.
>
>
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Grant
> Bagdasarian
> *Sent:* Tuesday, June 28, 2016 6:40 AM
> *To:* FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org) <
> freeswitch-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] FS not relaying 180/183 in proxy_media
>
>
>
> Hello,
>
>
>
> Is it possible to make freeswitch relay the 180 or 183 replies, including
> the ringback tone, in proxy_media mode to the next hop?
>
>
>
> Regards,
>
>
>
> Grant
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From s.safarov at gmail.com  Tue Jun 28 18:44:01 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 28 Jun 2016 14:44:01 +0000
Subject: [Freeswitch-users] FS not relaying 180/183 in proxy_media
In-Reply-To: <CAHtxdDc4x3gbsW6eGRzWRbeYUxHonwxWBb1WFDiWAH4P+TaipQ@mail.gmail.com>
References: <ada0d1fd3f3c4ea389d28a40cb26bb9f@CM-EX-V01.cm.local>
	<89d501d1d14a$25a795e0$70f6c1a0$@freeswitch.org>
	<CAHtxdDc4x3gbsW6eGRzWRbeYUxHonwxWBb1WFDiWAH4P+TaipQ@mail.gmail.com>
Message-ID: <CAHtxdDf204Q2nwqMW6emd911-0+6M7eOfkKrov-PNs2+tj_WRw@mail.gmail.com>

ZRTP in paththru mode.

On Tue, Jun 28, 2016, 17:41 Sergey Safarov <s.safarov at gmail.com> wrote:

> Current FS requires proxy media if you use zrtp and exist NAT in media
> path.
>
> On Tue, Jun 28, 2016, 17:35 Ken Rice <krice at freeswitch.org> wrote:
>
>> FS auto-relays those in any mode? however proxy media itself has no real
>> advantage over just setting late negotiation which reduces or eliminates
>> transcoding. Transcoding is where most of the load is there? as far as the
>> 180/183 as long as its 1:1 call in to call out from the FS perspective
>> those are passed across the bridge automatically? if you are doing
>> something like get a call in then originate another call and trying to
>> manually bridge them after the b leg is answered, this will not work and is
>> recommended against generally.
>>
>>
>>
>>
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Grant
>> Bagdasarian
>> *Sent:* Tuesday, June 28, 2016 6:40 AM
>> *To:* FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org) <
>> freeswitch-users at lists.freeswitch.org>
>> *Subject:* [Freeswitch-users] FS not relaying 180/183 in proxy_media
>>
>>
>>
>> Hello,
>>
>>
>>
>> Is it possible to make freeswitch relay the 180 or 183 replies, including
>> the ringback tone, in proxy_media mode to the next hop?
>>
>>
>>
>> Regards,
>>
>>
>>
>> Grant
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
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From john.nash778 at gmail.com  Tue Jun 28 18:46:04 2016
From: john.nash778 at gmail.com (John Nash)
Date: Tue, 28 Jun 2016 20:16:04 +0530
Subject: [Freeswitch-users] wss listener not starting
In-Reply-To: <CAND18T21EhRANNkfKaVwQZODsByf3JmtChZ8iW1p3bp8aXV+3A@mail.gmail.com>
References: <CAA4rGPxVFE=949-Q8c=RbByWhishu6fqztL31tLx7xBLc1BtaA@mail.gmail.com>
	<CAND18T21EhRANNkfKaVwQZODsByf3JmtChZ8iW1p3bp8aXV+3A@mail.gmail.com>
Message-ID: <CAA4rGPy2sFS1Wj0GNQrC6ze0F8YtFxOBtNrH0S3c06k6AJLhzA@mail.gmail.com>

It says WSS-BIND-URL            sips:mod_sofia@<my ip>:4431;transport=wss

On Tue, Jun 28, 2016 at 8:07 PM, Mirko Brankovic <mirkobrankovic at gmail.com>
wrote:

> Hi John,
>
> I guess all you need is:
>     <param name="wss-binding" value=":<your_port>"/>
> and proper wss.pem in certs/ dir
>
> What does  WSS-BIND-URL says on sofia status profile <your_new_profile>,
> if it starts at all ?
>
>
> On Tue, Jun 28, 2016 at 4:16 PM, John Nash <john.nash778 at gmail.com> wrote:
>
>> I have been using freeswitch for udp and today I created a new profile
>> and uncommented wss line and configured as per
>> https://freeswitch.org/confluence/display/FREESWITCH/WebRTC
>>
>> But after i rstarted and tried to check sofia status I do not see wss
>> listener (I see udp and TLS) In log also I do not see any error or anything
>> related to wss (INFO level)
>>
>> Am I missing somthing obvious?...any module dependency for wss?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Regards,
> Mirko
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mirkobrankovic at gmail.com  Tue Jun 28 18:54:19 2016
From: mirkobrankovic at gmail.com (Mirko Brankovic)
Date: Tue, 28 Jun 2016 16:54:19 +0200
Subject: [Freeswitch-users] wss listener not starting
In-Reply-To: <CAA4rGPy2sFS1Wj0GNQrC6ze0F8YtFxOBtNrH0S3c06k6AJLhzA@mail.gmail.com>
References: <CAA4rGPxVFE=949-Q8c=RbByWhishu6fqztL31tLx7xBLc1BtaA@mail.gmail.com>
	<CAND18T21EhRANNkfKaVwQZODsByf3JmtChZ8iW1p3bp8aXV+3A@mail.gmail.com>
	<CAA4rGPy2sFS1Wj0GNQrC6ze0F8YtFxOBtNrH0S3c06k6AJLhzA@mail.gmail.com>
Message-ID: <CAND18T1Be_RiKWii-XwM3NvxeidR5w87S=16hu8OJQeyO-oZZg@mail.gmail.com>

And what does:

On Tue, Jun 28, 2016 at 4:46 PM, John Nash <john.nash778 at gmail.com> wrote:

> It says WSS-BIND-URL            sips:mod_sofia@<my ip>:4431;transport=wss
>
> On Tue, Jun 28, 2016 at 8:07 PM, Mirko Brankovic <mirkobrankovic at gmail.com
> > wrote:
>
>> Hi John,
>>
>> I guess all you need is:
>>     <param name="wss-binding" value=":<your_port>"/>
>> and proper wss.pem in certs/ dir
>>
>> What does  WSS-BIND-URL says on sofia status profile <your_new_profile>,
>> if it starts at all ?
>>
>>
>> On Tue, Jun 28, 2016 at 4:16 PM, John Nash <john.nash778 at gmail.com>
>> wrote:
>>
>>> I have been using freeswitch for udp and today I created a new profile
>>> and uncommented wss line and configured as per
>>> https://freeswitch.org/confluence/display/FREESWITCH/WebRTC
>>>
>>> But after i rstarted and tried to check sofia status I do not see wss
>>> listener (I see udp and TLS) In log also I do not see any error or anything
>>> related to wss (INFO level)
>>>
>>> Am I missing somthing obvious?...any module dependency for wss?
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Regards,
>> Mirko
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Regards,
Mirko
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From mirkobrankovic at gmail.com  Tue Jun 28 18:54:53 2016
From: mirkobrankovic at gmail.com (Mirko Brankovic)
Date: Tue, 28 Jun 2016 16:54:53 +0200
Subject: [Freeswitch-users] wss listener not starting
In-Reply-To: <CAND18T1Be_RiKWii-XwM3NvxeidR5w87S=16hu8OJQeyO-oZZg@mail.gmail.com>
References: <CAA4rGPxVFE=949-Q8c=RbByWhishu6fqztL31tLx7xBLc1BtaA@mail.gmail.com>
	<CAND18T21EhRANNkfKaVwQZODsByf3JmtChZ8iW1p3bp8aXV+3A@mail.gmail.com>
	<CAA4rGPy2sFS1Wj0GNQrC6ze0F8YtFxOBtNrH0S3c06k6AJLhzA@mail.gmail.com>
	<CAND18T1Be_RiKWii-XwM3NvxeidR5w87S=16hu8OJQeyO-oZZg@mail.gmail.com>
Message-ID: <CAND18T17dfdvUcduRHMdj77Lpu2bqnMm9f85aC-s_=CwkQ2dfQ@mail.gmail.com>

Sorry,
and what does:
netstat -tlnp | grep 4431
says?

On Tue, Jun 28, 2016 at 4:54 PM, Mirko Brankovic <mirkobrankovic at gmail.com>
wrote:

> And what does:
>
> On Tue, Jun 28, 2016 at 4:46 PM, John Nash <john.nash778 at gmail.com> wrote:
>
>> It says WSS-BIND-URL            sips:mod_sofia@<my ip>:4431;transport=wss
>>
>> On Tue, Jun 28, 2016 at 8:07 PM, Mirko Brankovic <
>> mirkobrankovic at gmail.com> wrote:
>>
>>> Hi John,
>>>
>>> I guess all you need is:
>>>     <param name="wss-binding" value=":<your_port>"/>
>>> and proper wss.pem in certs/ dir
>>>
>>> What does  WSS-BIND-URL says on sofia status profile <your_new_profile>,
>>> if it starts at all ?
>>>
>>>
>>> On Tue, Jun 28, 2016 at 4:16 PM, John Nash <john.nash778 at gmail.com>
>>> wrote:
>>>
>>>> I have been using freeswitch for udp and today I created a new profile
>>>> and uncommented wss line and configured as per
>>>> https://freeswitch.org/confluence/display/FREESWITCH/WebRTC
>>>>
>>>> But after i rstarted and tried to check sofia status I do not see wss
>>>> listener (I see udp and TLS) In log also I do not see any error or anything
>>>> related to wss (INFO level)
>>>>
>>>> Am I missing somthing obvious?...any module dependency for wss?
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Regards,
>>> Mirko
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Regards,
> Mirko
>



-- 
Regards,
Mirko
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From gb at cm.nl  Tue Jun 28 19:09:22 2016
From: gb at cm.nl (Grant Bagdasarian)
Date: Tue, 28 Jun 2016 15:09:22 +0000
Subject: [Freeswitch-users] FS not relaying 180/183 in proxy_media
In-Reply-To: <89d501d1d14a$25a795e0$70f6c1a0$@freeswitch.org>
References: <ada0d1fd3f3c4ea389d28a40cb26bb9f@CM-EX-V01.cm.local>
	<89d501d1d14a$25a795e0$70f6c1a0$@freeswitch.org>
Message-ID: <cfb33fef038f44919572c1551d73f0d3@CM-EX-V01.cm.local>

I've enabled inbound-late-negotiation in the sip_profile and here is my dialplan config:

    <param name="inbound-bypass-media" value="false"/>
    <param name="inbound-proxy-media" value="true"/>
<param name="inbound-late-negotiation" value="true"/>

For now I've set the ring_ready, else no ring back tone is generated.

Anything wrong here?

<include>
    <context name="local">
    <extension name="route_to_public">
      <condition field="destination_number" expression="(\d+)$" require-nested="false">
        <action application="set" data="bridge_early_media=true" />
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=false"/>
                               <action application="set" data="ignore_early_media=false"/>
        <action application="set" data="sip_copy_custom_headers=false" />
                               <action application="set" data="sip_ignore_reinvites=false"/>

                               <action application="ring_ready" />

                               <action application="set" data="effective_caller_id_number=${translate(${caller_id_number} To_International)}" />
        <action application="set" data="effective_caller_id_name=${translate(${caller_id_number} To_International)}" />
                               <action application="set" data="destination_number_translated=${translate(${destination_number} To_International)}"/>
        <action application="bridge" data="sofia/gateway/${distributor(public_gws ${sofia(profile pubilc_eth1 gwlist down)})}/${destination_number_translated}" />
      </condition>
    </extension>
  </context>
</include>


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: Tuesday, June 28, 2016 4:34 PM
To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] FS not relaying 180/183 in proxy_media

FS auto-relays those in any mode... however proxy media itself has no real advantage over just setting late negotiation which reduces or eliminates transcoding. Transcoding is where most of the load is there... as far as the 180/183 as long as its 1:1 call in to call out from the FS perspective those are passed across the bridge automatically... if you are doing something like get a call in then originate another call and trying to manually bridge them after the b leg is answered, this will not work and is recommended against generally.



From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Grant Bagdasarian
Sent: Tuesday, June 28, 2016 6:40 AM
To: FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>) <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: [Freeswitch-users] FS not relaying 180/183 in proxy_media

Hello,

Is it possible to make freeswitch relay the 180 or 183 replies, including the ringback tone, in proxy_media mode to the next hop?

Regards,

Grant
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From john.nash778 at gmail.com  Tue Jun 28 19:09:20 2016
From: john.nash778 at gmail.com (John Nash)
Date: Tue, 28 Jun 2016 20:39:20 +0530
Subject: [Freeswitch-users] wss listener not starting
In-Reply-To: <CAND18T17dfdvUcduRHMdj77Lpu2bqnMm9f85aC-s_=CwkQ2dfQ@mail.gmail.com>
References: <CAA4rGPxVFE=949-Q8c=RbByWhishu6fqztL31tLx7xBLc1BtaA@mail.gmail.com>
	<CAND18T21EhRANNkfKaVwQZODsByf3JmtChZ8iW1p3bp8aXV+3A@mail.gmail.com>
	<CAA4rGPy2sFS1Wj0GNQrC6ze0F8YtFxOBtNrH0S3c06k6AJLhzA@mail.gmail.com>
	<CAND18T1Be_RiKWii-XwM3NvxeidR5w87S=16hu8OJQeyO-oZZg@mail.gmail.com>
	<CAND18T17dfdvUcduRHMdj77Lpu2bqnMm9f85aC-s_=CwkQ2dfQ@mail.gmail.com>
Message-ID: <CAA4rGPxiuz7NMMUMefVOoNnx-vP0xdDEm-Xw0jOeLbzr668OSw@mail.gmail.com>

Sorry should have checked this before. As per netstat -tulpn
<http://www.cyberciti.biz/tips/linux-display-open-ports-owner.html>
freeswitch is listening to 4431 port. I also checked brwoser log (Using
sip.js very simple code to register) in Browser I saw Method not allowed
and then occured to me that may be I have not allowed register in the
profile and that was it !!!

Now I will try some tests and see how it goes.

On Tue, Jun 28, 2016 at 8:24 PM, Mirko Brankovic <mirkobrankovic at gmail.com>
wrote:

> Sorry,
> and what does:
> netstat -tlnp | grep 4431
> says?
>
> On Tue, Jun 28, 2016 at 4:54 PM, Mirko Brankovic <mirkobrankovic at gmail.com
> > wrote:
>
>> And what does:
>>
>> On Tue, Jun 28, 2016 at 4:46 PM, John Nash <john.nash778 at gmail.com>
>> wrote:
>>
>>> It says WSS-BIND-URL            sips:mod_sofia@<my
>>> ip>:4431;transport=wss
>>>
>>> On Tue, Jun 28, 2016 at 8:07 PM, Mirko Brankovic <
>>> mirkobrankovic at gmail.com> wrote:
>>>
>>>> Hi John,
>>>>
>>>> I guess all you need is:
>>>>     <param name="wss-binding" value=":<your_port>"/>
>>>> and proper wss.pem in certs/ dir
>>>>
>>>> What does  WSS-BIND-URL says on sofia status profile
>>>> <your_new_profile>, if it starts at all ?
>>>>
>>>>
>>>> On Tue, Jun 28, 2016 at 4:16 PM, John Nash <john.nash778 at gmail.com>
>>>> wrote:
>>>>
>>>>> I have been using freeswitch for udp and today I created a new profile
>>>>> and uncommented wss line and configured as per
>>>>> https://freeswitch.org/confluence/display/FREESWITCH/WebRTC
>>>>>
>>>>> But after i rstarted and tried to check sofia status I do not see wss
>>>>> listener (I see udp and TLS) In log also I do not see any error or anything
>>>>> related to wss (INFO level)
>>>>>
>>>>> Am I missing somthing obvious?...any module dependency for wss?
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Regards,
>>>> Mirko
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Regards,
>> Mirko
>>
>
>
>
> --
> Regards,
> Mirko
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mirkobrankovic at gmail.com  Tue Jun 28 19:18:52 2016
From: mirkobrankovic at gmail.com (Mirko Brankovic)
Date: Tue, 28 Jun 2016 17:18:52 +0200
Subject: [Freeswitch-users] wss listener not starting
In-Reply-To: <CAA4rGPxiuz7NMMUMefVOoNnx-vP0xdDEm-Xw0jOeLbzr668OSw@mail.gmail.com>
References: <CAA4rGPxVFE=949-Q8c=RbByWhishu6fqztL31tLx7xBLc1BtaA@mail.gmail.com>
	<CAND18T21EhRANNkfKaVwQZODsByf3JmtChZ8iW1p3bp8aXV+3A@mail.gmail.com>
	<CAA4rGPy2sFS1Wj0GNQrC6ze0F8YtFxOBtNrH0S3c06k6AJLhzA@mail.gmail.com>
	<CAND18T1Be_RiKWii-XwM3NvxeidR5w87S=16hu8OJQeyO-oZZg@mail.gmail.com>
	<CAND18T17dfdvUcduRHMdj77Lpu2bqnMm9f85aC-s_=CwkQ2dfQ@mail.gmail.com>
	<CAA4rGPxiuz7NMMUMefVOoNnx-vP0xdDEm-Xw0jOeLbzr668OSw@mail.gmail.com>
Message-ID: <CAND18T1vE-LcLrAsRuocpxdY4N2Lo1TdVvQY-7fhDfjbzHneGA@mail.gmail.com>

Np :)
You can test your socket with this:
https://www.websocket.org/echo.html

On Tue, Jun 28, 2016 at 5:09 PM, John Nash <john.nash778 at gmail.com> wrote:

> Sorry should have checked this before. As per netstat -tulpn
> <http://www.cyberciti.biz/tips/linux-display-open-ports-owner.html>
> freeswitch is listening to 4431 port. I also checked brwoser log (Using
> sip.js very simple code to register) in Browser I saw Method not allowed
> and then occured to me that may be I have not allowed register in the
> profile and that was it !!!
>
> Now I will try some tests and see how it goes.
>
> On Tue, Jun 28, 2016 at 8:24 PM, Mirko Brankovic <mirkobrankovic at gmail.com
> > wrote:
>
>> Sorry,
>> and what does:
>> netstat -tlnp | grep 4431
>> says?
>>
>> On Tue, Jun 28, 2016 at 4:54 PM, Mirko Brankovic <
>> mirkobrankovic at gmail.com> wrote:
>>
>>> And what does:
>>>
>>> On Tue, Jun 28, 2016 at 4:46 PM, John Nash <john.nash778 at gmail.com>
>>> wrote:
>>>
>>>> It says WSS-BIND-URL            sips:mod_sofia@<my
>>>> ip>:4431;transport=wss
>>>>
>>>> On Tue, Jun 28, 2016 at 8:07 PM, Mirko Brankovic <
>>>> mirkobrankovic at gmail.com> wrote:
>>>>
>>>>> Hi John,
>>>>>
>>>>> I guess all you need is:
>>>>>     <param name="wss-binding" value=":<your_port>"/>
>>>>> and proper wss.pem in certs/ dir
>>>>>
>>>>> What does  WSS-BIND-URL says on sofia status profile
>>>>> <your_new_profile>, if it starts at all ?
>>>>>
>>>>>
>>>>> On Tue, Jun 28, 2016 at 4:16 PM, John Nash <john.nash778 at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> I have been using freeswitch for udp and today I created a new
>>>>>> profile and uncommented wss line and configured as per
>>>>>> https://freeswitch.org/confluence/display/FREESWITCH/WebRTC
>>>>>>
>>>>>> But after i rstarted and tried to check sofia status I do not see wss
>>>>>> listener (I see udp and TLS) In log also I do not see any error or anything
>>>>>> related to wss (INFO level)
>>>>>>
>>>>>> Am I missing somthing obvious?...any module dependency for wss?
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Regards,
>>>>> Mirko
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Regards,
>>> Mirko
>>>
>>
>>
>>
>> --
>> Regards,
>> Mirko
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Regards,
Mirko
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From bipin at xbipin.com  Tue Jun 28 19:33:18 2016
From: bipin at xbipin.com (Bipin Patel)
Date: Tue, 28 Jun 2016 19:33:18 +0400
Subject: [Freeswitch-users] any1 using opus in zoiper with FS
 successfully?
In-Reply-To: <229f2220-82ae-4a85-7613-ed762e9914a3@xbipin.com>
References: <229f2220-82ae-4a85-7613-ed762e9914a3@xbipin.com>
Message-ID: <1d720381-b0b5-c11d-c98f-da93b7095584@xbipin.com>

i got a reply from zoiper devs and they r blaming it on FS

"We have done some test with Freeswitch version 1.4.26 and there are no 
problems with Opus. Most probably something has been changed in the 
newer versions of Freeswitch."



Regards,
Bipin



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

-------- Original Message  --------
Subject: [Freeswitch-users] any1 using opus in zoiper with FS successfully?
From: Bipin Patel <bipin at xbipin.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date: 6/17/2016, 8:03:00 PM

> is there any1 using opus codec with FS using zoiper on android, coz im
> banging my head from past few days but calling from zoiper to zoiper
> always gives one way audio even though the other way RTP reaches FS but
> it doesnt send it back to zoiper and thats the cause of one way audio. I
> tried playing with the opus settings in almost all combinations but it
> just doesnt work, my profile has late negotiate and inherit codec
> enabled, tried disabling inherit codec but its still the same. FS cli
> also doesnt show any error regarding y it isnt sending the RTP back to
> dialer even though it gets it from the other zoiper
>
> the packet trace is there on jira FS-9270 if any1 wants to investigate,
> any help is appreciated
>



From john.nash778 at gmail.com  Tue Jun 28 19:44:40 2016
From: john.nash778 at gmail.com (John Nash)
Date: Tue, 28 Jun 2016 21:14:40 +0530
Subject: [Freeswitch-users] wss listener not starting
In-Reply-To: <CAND18T1vE-LcLrAsRuocpxdY4N2Lo1TdVvQY-7fhDfjbzHneGA@mail.gmail.com>
References: <CAA4rGPxVFE=949-Q8c=RbByWhishu6fqztL31tLx7xBLc1BtaA@mail.gmail.com>
	<CAND18T21EhRANNkfKaVwQZODsByf3JmtChZ8iW1p3bp8aXV+3A@mail.gmail.com>
	<CAA4rGPy2sFS1Wj0GNQrC6ze0F8YtFxOBtNrH0S3c06k6AJLhzA@mail.gmail.com>
	<CAND18T1Be_RiKWii-XwM3NvxeidR5w87S=16hu8OJQeyO-oZZg@mail.gmail.com>
	<CAND18T17dfdvUcduRHMdj77Lpu2bqnMm9f85aC-s_=CwkQ2dfQ@mail.gmail.com>
	<CAA4rGPxiuz7NMMUMefVOoNnx-vP0xdDEm-Xw0jOeLbzr668OSw@mail.gmail.com>
	<CAND18T1vE-LcLrAsRuocpxdY4N2Lo1TdVvQY-7fhDfjbzHneGA@mail.gmail.com>
Message-ID: <CAA4rGPzWwsfhE7sz_Z=OmqAV0QMELFxNAAQHi7byib8kf66H_w@mail.gmail.com>

Cool

On Tue, Jun 28, 2016 at 8:48 PM, Mirko Brankovic <mirkobrankovic at gmail.com>
wrote:

> Np :)
> You can test your socket with this:
> https://www.websocket.org/echo.html
>
> On Tue, Jun 28, 2016 at 5:09 PM, John Nash <john.nash778 at gmail.com> wrote:
>
>> Sorry should have checked this before. As per netstat -tulpn
>> <http://www.cyberciti.biz/tips/linux-display-open-ports-owner.html>
>> freeswitch is listening to 4431 port. I also checked brwoser log (Using
>> sip.js very simple code to register) in Browser I saw Method not allowed
>> and then occured to me that may be I have not allowed register in the
>> profile and that was it !!!
>>
>> Now I will try some tests and see how it goes.
>>
>> On Tue, Jun 28, 2016 at 8:24 PM, Mirko Brankovic <
>> mirkobrankovic at gmail.com> wrote:
>>
>>> Sorry,
>>> and what does:
>>> netstat -tlnp | grep 4431
>>> says?
>>>
>>> On Tue, Jun 28, 2016 at 4:54 PM, Mirko Brankovic <
>>> mirkobrankovic at gmail.com> wrote:
>>>
>>>> And what does:
>>>>
>>>> On Tue, Jun 28, 2016 at 4:46 PM, John Nash <john.nash778 at gmail.com>
>>>> wrote:
>>>>
>>>>> It says WSS-BIND-URL            sips:mod_sofia@<my
>>>>> ip>:4431;transport=wss
>>>>>
>>>>> On Tue, Jun 28, 2016 at 8:07 PM, Mirko Brankovic <
>>>>> mirkobrankovic at gmail.com> wrote:
>>>>>
>>>>>> Hi John,
>>>>>>
>>>>>> I guess all you need is:
>>>>>>     <param name="wss-binding" value=":<your_port>"/>
>>>>>> and proper wss.pem in certs/ dir
>>>>>>
>>>>>> What does  WSS-BIND-URL says on sofia status profile
>>>>>> <your_new_profile>, if it starts at all ?
>>>>>>
>>>>>>
>>>>>> On Tue, Jun 28, 2016 at 4:16 PM, John Nash <john.nash778 at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>> I have been using freeswitch for udp and today I created a new
>>>>>>> profile and uncommented wss line and configured as per
>>>>>>> https://freeswitch.org/confluence/display/FREESWITCH/WebRTC
>>>>>>>
>>>>>>> But after i rstarted and tried to check sofia status I do not see
>>>>>>> wss listener (I see udp and TLS) In log also I do not see any error or
>>>>>>> anything related to wss (INFO level)
>>>>>>>
>>>>>>> Am I missing somthing obvious?...any module dependency for wss?
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Regards,
>>>>>> Mirko
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Regards,
>>>> Mirko
>>>>
>>>
>>>
>>>
>>> --
>>> Regards,
>>> Mirko
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Regards,
> Mirko
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From luis.azedo at factorlusitano.com  Tue Jun 28 22:06:27 2016
From: luis.azedo at factorlusitano.com (Luis Azedo)
Date: Tue, 28 Jun 2016 19:06:27 +0100
Subject: [Freeswitch-users] bind_digits_action regexp
Message-ID: <CAEvFzR_ze4FZpQcFGt3R9nM4fVUMteqoO-rDeJR=58rPh=ox+w@mail.gmail.com>

Hi,

i'm trying to collect as many digits as possible but only getting the first.

bind_digit_action(my_domain,'~^\*(\d+)$',exec:execute_extension,MY_EXTENSION
XML my_context,both,self)
if i press *8754, MY_EXTENSION is executed as soon as i type the 8. i guess
the regexp is true, but shouldn't it try to capture the rest of the digits ?

also tried
bind_digit_action(my_domain,'~^\*(\d{5,12})$',exec:execute_extension,MY_EXTENSION
XML my_context,both,self)
when i press *123456789, MY_EXTENSION is executed as soon as i type the 5

also used 'set(bind_digit_input_timeout=15000)' with no luck
is there a way to capture until bind_digit_input_timeout ?

Thanks
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From david.ponzone at gmail.com  Wed Jun 29 01:09:19 2016
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 28 Jun 2016 23:09:19 +0200
Subject: [Freeswitch-users] bind_digits_action regexp
In-Reply-To: <CAEvFzR_ze4FZpQcFGt3R9nM4fVUMteqoO-rDeJR=58rPh=ox+w@mail.gmail.com>
References: <CAEvFzR_ze4FZpQcFGt3R9nM4fVUMteqoO-rDeJR=58rPh=ox+w@mail.gmail.com>
Message-ID: <8B88F2B3-4E13-4ECD-B726-E800C7370644@gmail.com>

According the wiki page, it cannot be used for capture.

David Ponzone



> Le 28 juin 2016 ? 20:06, Luis Azedo <luis.azedo at factorlusitano.com> a ?crit :
> 
> Hi,
> 
> i'm trying to collect as many digits as possible but only getting the first.
> 
> bind_digit_action(my_domain,'~^\*(\d+)$',exec:execute_extension,MY_EXTENSION XML my_context,both,self)
> if i press *8754, MY_EXTENSION is executed as soon as i type the 8. i guess the regexp is true, but shouldn't it try to capture the rest of the digits ?
> 
> also tried
> bind_digit_action(my_domain,'~^\*(\d{5,12})$',exec:execute_extension,MY_EXTENSION XML my_context,both,self)
> when i press *123456789, MY_EXTENSION is executed as soon as i type the 5
> 
> also used 'set(bind_digit_input_timeout=15000)' with no luck
> is there a way to capture until bind_digit_input_timeout ?
> 
> Thanks
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From aubalde at presenceco.com  Wed Jun 29 12:20:49 2016
From: aubalde at presenceco.com (=?UTF-8?Q?Agust=C3=AD_Ubalde?=)
Date: Wed, 29 Jun 2016 10:20:49 +0200
Subject: [Freeswitch-users] High CPU consumption
Message-ID: <CAKThY07a4Jpnbbtv9C1kin09eZ9DJYRMXHmp34dK=kwOWBi5-A@mail.gmail.com>

Hi all,

I have a high CPU consumption by FreeSWITCH. In fact, consumption is
highest on CPU system. On this server we have registered 20 sessions
(WebRTC).

Environment:

   - CentOS 6.7 64b
   - FreeSWITCh 1.5.14


The results of the execution of:

step 1) *ps -LlFm -p `pidof freeswitch`*

F S UID        PID  PPID   LWP  C NLWP PRI  NI ADDR SZ WCHAN    RSS PSR
STIME TTY          TIME CMD
1 - 499       4481     1     - 99   55   -   - - 216700 -     140840  -
Jun28 ?        17:55:10 /usr/bin/freeswitch -nc
1 S 499          -     -  4481  0    -  99  19 -     - poll_s     -   0
Jun28 -        00:00:00 -
1 S 499          -     -  4482  0    -  99  19 -     - poll_s     -   0
Jun28 -        00:00:00 -
1 S 499          -     -  4483  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:32 -
1 S 499          -     -  4484  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:03 -
1 S 499          -     -  4487  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:01 -
1 S 499          -     -  4488  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:04 -
1 S 499          -     -  4489  0    -  99  19 -     - poll_s     -   0
Jun28 -        00:00:00 -
1 S 499          -     -  4490  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:01 -
1 S 499          -     -  4492  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:00 -
1 S 499          -     -  4493  0    -  99  19 -     - ep_pol     -   0
Jun28 -        00:00:00 -
1 S 499          -     -  4494  0    -  99  19 -     - ep_pol     -   0
Jun28 -        00:00:00 -
*1 R 499          -     -  4495 97    -  99  19 -     - -          -   0
Jun28 -        17:37:34 -*
1 S 499          -     -  4496  0    -  99  19 -     - ep_pol     -   0
Jun28 -        00:00:00 -
1 S 499          -     -  4497  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:00 -
1 S 499          -     -  4498  0    -  99  19 -     - poll_s     -   0
Jun28 -        00:00:00 -
1 S 499          -     -  4499  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:00 -
1 S 499          -     -  4500  0    -  99  19 -     - poll_s     -   0
Jun28 -        00:00:01 -
1 S 499          -     -  4501  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:00 -
1 S 499          -     -  4502  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:00 -
1 S 499          -     -  4503  0    -  99  19 -     - poll_s     -   0
Jun28 -        00:00:00 -
1 S 499          -     -  4504  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:00 -
1 S 499          -     -  4510  0    -  99  19 -     - inet_c     -   0
Jun28 -        00:00:00 -
1 S 499          -     -  4511  0    -  99  19 -     - timerf     -   0
Jun28 -        00:00:18 -
1 S 499          -     -  4512  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:00 -
1 S 499          -     -  4513  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:00 -
1 S 499          -     -  4536  0    -  99  19 -     - hrtime     -   0
Jun28 -        00:00:03 -
1 S 499          -     -  4558  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:43 -
1 S 499          -     -  4559  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:43 -
1 S 499          -     -  4560  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:43 -
1 S 499          -     -  4561  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:43 -
1 S 499          -     -  4562  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:43 -
1 S 499          -     -  4564  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:39 -
1 S 499          -     -  4565  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:43 -
1 S 499          -     -  4567  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:39 -
1 S 499          -     -  4568  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:40 -
1 S 499          -     -  4569  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:40 -
1 S 499          -     -  4570  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:40 -
1 S 499          -     -  4571  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:39 -
1 S 499          -     -  4572  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:40 -
1 S 499          -     -  4573  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:39 -
1 S 499          -     -  4575  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:40 -
1 S 499          -     -  4636  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:17 -
1 S 499          -     -  4637  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:17 -
1 S 499          -     -  4638  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:17 -
1 S 499          -     -  4639  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:17 -
1 S 499          -     -  4640  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:17 -
1 S 499          -     -  4641  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:17 -
1 S 499          -     -  4642  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:17 -
1 S 499          -     -  4643  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:17 -
1 S 499          -     -  4644  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:17 -
1 S 499          -     -  4645  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:17 -
1 S 499          -     -  4646  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:17 -
1 S 499          -     -  4647  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:17 -
1 S 499          -     -  4648  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:17 -
1 S 499          -     -  4649  0    -  99  19 -     - futex_     -   0
Jun28 -        00:00:17 -

step 2) *pstack 4481*

Thread 45 (Thread 0x7ff6ba08a700 (LWP 4495)):
#0  0x0000003410ce8e63 in epoll_wait () from /lib64/libc.so.6
#1  0x00007ff6b877bba5 in ?? () from /usr/local/freeswitch/mod/mod_sofia.so
#2  0x00007ff6b877ba28 in ?? () from /usr/local/freeswitch/mod/mod_sofia.so
#3  0x00007ff6b8779715 in ?? () from /usr/local/freeswitch/mod/mod_sofia.so
#4  0x00000034110079d1 in start_thread () from /lib64/libpthread.so.0
#5  0x0000003410ce886d in clone () from /lib64/libc.so.6

Any idea?


Thanks,

*PRESENCE TECHNOLOGY*
*Agust? Ubalde Bellot*
Chief Developer
C/ Comte Urgell 240 3A
Barcelona 08036
aubalde at presenceco.com

Ph: +34 93 10 10 300
Fx: +34 93 10 10 333

*www.presenceco.com* <http://www.presenceco.com/>

*Follow us on:*

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-- 



*Presence Technology - DisclaimerThis message, its content and any file 
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/or privileged. If you have received this e-mail in error or had access to 
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From gmaruzz at gmail.com  Wed Jun 29 12:27:10 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 29 Jun 2016 10:27:10 +0200
Subject: [Freeswitch-users] High CPU consumption
In-Reply-To: <CAKThY07a4Jpnbbtv9C1kin09eZ9DJYRMXHmp34dK=kwOWBi5-A@mail.gmail.com>
References: <CAKThY07a4Jpnbbtv9C1kin09eZ9DJYRMXHmp34dK=kwOWBi5-A@mail.gmail.com>
Message-ID: <CALXCt0r_bmCnmQqVyvTuvJ2yMDp=Hc_0hf86VmSw5u=mj_eaUg@mail.gmail.com>

Video muxing is heavy on CPU, that's normal.

Also, you will find best performances and reliability using Debian 8
Jessie, and latest FreeSWITCH release.

Your specific version of FreeSWITCH is too old.

CentOS 6 is not supposed to be used for video (webrtc) service.

-giovanni

On Wed, Jun 29, 2016 at 10:20 AM, Agust? Ubalde <aubalde at presenceco.com>
wrote:

> Hi all,
>
> I have a high CPU consumption by FreeSWITCH. In fact, consumption is
> highest on CPU system. On this server we have registered 20 sessions
> (WebRTC).
>
> Environment:
>
>    - CentOS 6.7 64b
>    - FreeSWITCh 1.5.14
>
>
> The results of the execution of:
>
> step 1) *ps -LlFm -p `pidof freeswitch`*
>
> F S UID        PID  PPID   LWP  C NLWP PRI  NI ADDR SZ WCHAN    RSS PSR
> STIME TTY          TIME CMD
> 1 - 499       4481     1     - 99   55   -   - - 216700 -     140840  -
> Jun28 ?        17:55:10 /usr/bin/freeswitch -nc
> 1 S 499          -     -  4481  0    -  99  19 -     - poll_s     -   0
> Jun28 -        00:00:00 -
> 1 S 499          -     -  4482  0    -  99  19 -     - poll_s     -   0
> Jun28 -        00:00:00 -
> 1 S 499          -     -  4483  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:32 -
> 1 S 499          -     -  4484  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:03 -
> 1 S 499          -     -  4487  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:01 -
> 1 S 499          -     -  4488  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:04 -
> 1 S 499          -     -  4489  0    -  99  19 -     - poll_s     -   0
> Jun28 -        00:00:00 -
> 1 S 499          -     -  4490  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:01 -
> 1 S 499          -     -  4492  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:00 -
> 1 S 499          -     -  4493  0    -  99  19 -     - ep_pol     -   0
> Jun28 -        00:00:00 -
> 1 S 499          -     -  4494  0    -  99  19 -     - ep_pol     -   0
> Jun28 -        00:00:00 -
> *1 R 499          -     -  4495 97    -  99  19 -     - -          -   0
> Jun28 -        17:37:34 -*
> 1 S 499          -     -  4496  0    -  99  19 -     - ep_pol     -   0
> Jun28 -        00:00:00 -
> 1 S 499          -     -  4497  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:00 -
> 1 S 499          -     -  4498  0    -  99  19 -     - poll_s     -   0
> Jun28 -        00:00:00 -
> 1 S 499          -     -  4499  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:00 -
> 1 S 499          -     -  4500  0    -  99  19 -     - poll_s     -   0
> Jun28 -        00:00:01 -
> 1 S 499          -     -  4501  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:00 -
> 1 S 499          -     -  4502  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:00 -
> 1 S 499          -     -  4503  0    -  99  19 -     - poll_s     -   0
> Jun28 -        00:00:00 -
> 1 S 499          -     -  4504  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:00 -
> 1 S 499          -     -  4510  0    -  99  19 -     - inet_c     -   0
> Jun28 -        00:00:00 -
> 1 S 499          -     -  4511  0    -  99  19 -     - timerf     -   0
> Jun28 -        00:00:18 -
> 1 S 499          -     -  4512  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:00 -
> 1 S 499          -     -  4513  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:00 -
> 1 S 499          -     -  4536  0    -  99  19 -     - hrtime     -   0
> Jun28 -        00:00:03 -
> 1 S 499          -     -  4558  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:43 -
> 1 S 499          -     -  4559  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:43 -
> 1 S 499          -     -  4560  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:43 -
> 1 S 499          -     -  4561  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:43 -
> 1 S 499          -     -  4562  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:43 -
> 1 S 499          -     -  4564  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:39 -
> 1 S 499          -     -  4565  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:43 -
> 1 S 499          -     -  4567  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:39 -
> 1 S 499          -     -  4568  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:40 -
> 1 S 499          -     -  4569  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:40 -
> 1 S 499          -     -  4570  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:40 -
> 1 S 499          -     -  4571  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:39 -
> 1 S 499          -     -  4572  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:40 -
> 1 S 499          -     -  4573  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:39 -
> 1 S 499          -     -  4575  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:40 -
> 1 S 499          -     -  4636  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:17 -
> 1 S 499          -     -  4637  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:17 -
> 1 S 499          -     -  4638  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:17 -
> 1 S 499          -     -  4639  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:17 -
> 1 S 499          -     -  4640  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:17 -
> 1 S 499          -     -  4641  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:17 -
> 1 S 499          -     -  4642  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:17 -
> 1 S 499          -     -  4643  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:17 -
> 1 S 499          -     -  4644  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:17 -
> 1 S 499          -     -  4645  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:17 -
> 1 S 499          -     -  4646  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:17 -
> 1 S 499          -     -  4647  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:17 -
> 1 S 499          -     -  4648  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:17 -
> 1 S 499          -     -  4649  0    -  99  19 -     - futex_     -   0
> Jun28 -        00:00:17 -
>
> step 2) *pstack 4481*
>
> Thread 45 (Thread 0x7ff6ba08a700 (LWP 4495)):
> #0  0x0000003410ce8e63 in epoll_wait () from /lib64/libc.so.6
> #1  0x00007ff6b877bba5 in ?? () from /usr/local/freeswitch/mod/mod_sofia.so
> #2  0x00007ff6b877ba28 in ?? () from /usr/local/freeswitch/mod/mod_sofia.so
> #3  0x00007ff6b8779715 in ?? () from /usr/local/freeswitch/mod/mod_sofia.so
> #4  0x00000034110079d1 in start_thread () from /lib64/libpthread.so.0
> #5  0x0000003410ce886d in clone () from /lib64/libc.so.6
>
> Any idea?
>
>
> Thanks,
>
> *PRESENCE TECHNOLOGY*
> *Agust? Ubalde Bellot*
> Chief Developer
> C/ Comte Urgell 240 3A
> Barcelona 08036
> aubalde at presenceco.com
>
> Ph: +34 93 10 10 300
> Fx: +34 93 10 10 333
>
> *www.presenceco.com* <http://www.presenceco.com/>
>
> *Follow us on:*
>
> *[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
> <http://www.youtube.com/PresenceTechnology>*[image: in]*
> <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]*
> <http://www.slideshare.net/PresenceTech>*[image: fb]*
> <http://www.facebook.com/presencetech>
>
> For additional information, please visit our website *www.presenceco.com*
> <http://www.presenceco.com/>
>
>
> *Presence Technology - DisclaimerThis message, its content and any file
> attached thereto is for the intended recipient only and is confidential and
> /or privileged. If you have received this e-mail in error or had access to
> it, you should note that the information in it is private and any use
> thereof is unauthorized. In such an event please notify us by e-mail or by
> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by
> whatsoever means and any transmission or dissemination thereof to other
> persons is prohibited. It should be deleted immediately from your system.
> Presence Technology reserves the right to take legal action against any
> persons unlawfully gaining access to the content of any external message it
> has emitted.*
>
> *For additional information, please visit our website **www.presenceco.com
> <http://www.presenceco.com/>*
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
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> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From aubalde at presenceco.com  Wed Jun 29 12:37:35 2016
From: aubalde at presenceco.com (=?UTF-8?Q?Agust=C3=AD_Ubalde?=)
Date: Wed, 29 Jun 2016 10:37:35 +0200
Subject: [Freeswitch-users] High CPU consumption
In-Reply-To: <CALXCt0r_bmCnmQqVyvTuvJ2yMDp=Hc_0hf86VmSw5u=mj_eaUg@mail.gmail.com>
References: <CAKThY07a4Jpnbbtv9C1kin09eZ9DJYRMXHmp34dK=kwOWBi5-A@mail.gmail.com>
	<CALXCt0r_bmCnmQqVyvTuvJ2yMDp=Hc_0hf86VmSw5u=mj_eaUg@mail.gmail.com>
Message-ID: <CAKThY040oQGR8e-5+f2fsDC2ArG_EVXke8iX975A5tk=eqJVBw@mail.gmail.com>

Hi Giovanni,

I don't use video muxing, only audio.

This is a more detailed stack:

Thread 45 (Thread 0x7ff6ba08a700 (LWP 4495)):
#0  0x0000003410ce8e63 in epoll_wait () from /lib64/libc.so.6
#1  0x00007ff6b877bba5 in su_epoll_port_wait_events () from
/usr/local/freeswitch/mod/mod_sofia.so
#2  0x00007ff6b877ba28 in su_base_port_run () from
/usr/local/freeswitch/mod/mod_sofia.so
#3  0x00007ff6b8779715 in su_pthread_port_clone_main () from
/usr/local/freeswitch/mod/mod_sofia.so
#4  0x00000034110079d1 in start_thread () from /lib64/libpthread.so.0
#5  0x0000003410ce886d in clone () from /lib64/libc.so.6


Thanks,

*PRESENCE TECHNOLOGY*
*Agust? Ubalde Bellot*
Chief Developer
C/ Comte Urgell 240 3A
Barcelona 08036
aubalde at presenceco.com

Ph: +34 93 10 10 300
Fx: +34 93 10 10 333

*www.presenceco.com* <http://www.presenceco.com/>

*Follow us on:*

*[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
<http://www.youtube.com/PresenceTechnology>*[image: in]*
<http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]*
<http://www.slideshare.net/PresenceTech>*[image: fb]*
<http://www.facebook.com/presencetech>

For additional information, please visit our website *www.presenceco.com*
<http://www.presenceco.com/>

2016-06-29 10:27 GMT+02:00 Giovanni Maruzzelli <gmaruzz at gmail.com>:

> Video muxing is heavy on CPU, that's normal.
>
> Also, you will find best performances and reliability using Debian 8
> Jessie, and latest FreeSWITCH release.
>
> Your specific version of FreeSWITCH is too old.
>
> CentOS 6 is not supposed to be used for video (webrtc) service.
>
> -giovanni
>
> On Wed, Jun 29, 2016 at 10:20 AM, Agust? Ubalde <aubalde at presenceco.com>
> wrote:
>
>> Hi all,
>>
>> I have a high CPU consumption by FreeSWITCH. In fact, consumption is
>> highest on CPU system. On this server we have registered 20 sessions
>> (WebRTC).
>>
>> Environment:
>>
>>    - CentOS 6.7 64b
>>    - FreeSWITCh 1.5.14
>>
>>
>> The results of the execution of:
>>
>> step 1) *ps -LlFm -p `pidof freeswitch`*
>>
>> F S UID        PID  PPID   LWP  C NLWP PRI  NI ADDR SZ WCHAN    RSS PSR
>> STIME TTY          TIME CMD
>> 1 - 499       4481     1     - 99   55   -   - - 216700 -     140840  -
>> Jun28 ?        17:55:10 /usr/bin/freeswitch -nc
>> 1 S 499          -     -  4481  0    -  99  19 -     - poll_s     -   0
>> Jun28 -        00:00:00 -
>> 1 S 499          -     -  4482  0    -  99  19 -     - poll_s     -   0
>> Jun28 -        00:00:00 -
>> 1 S 499          -     -  4483  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:32 -
>> 1 S 499          -     -  4484  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:03 -
>> 1 S 499          -     -  4487  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:01 -
>> 1 S 499          -     -  4488  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:04 -
>> 1 S 499          -     -  4489  0    -  99  19 -     - poll_s     -   0
>> Jun28 -        00:00:00 -
>> 1 S 499          -     -  4490  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:01 -
>> 1 S 499          -     -  4492  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:00 -
>> 1 S 499          -     -  4493  0    -  99  19 -     - ep_pol     -   0
>> Jun28 -        00:00:00 -
>> 1 S 499          -     -  4494  0    -  99  19 -     - ep_pol     -   0
>> Jun28 -        00:00:00 -
>> *1 R 499          -     -  4495 97    -  99  19 -     - -          -   0
>> Jun28 -        17:37:34 -*
>> 1 S 499          -     -  4496  0    -  99  19 -     - ep_pol     -   0
>> Jun28 -        00:00:00 -
>> 1 S 499          -     -  4497  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:00 -
>> 1 S 499          -     -  4498  0    -  99  19 -     - poll_s     -   0
>> Jun28 -        00:00:00 -
>> 1 S 499          -     -  4499  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:00 -
>> 1 S 499          -     -  4500  0    -  99  19 -     - poll_s     -   0
>> Jun28 -        00:00:01 -
>> 1 S 499          -     -  4501  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:00 -
>> 1 S 499          -     -  4502  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:00 -
>> 1 S 499          -     -  4503  0    -  99  19 -     - poll_s     -   0
>> Jun28 -        00:00:00 -
>> 1 S 499          -     -  4504  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:00 -
>> 1 S 499          -     -  4510  0    -  99  19 -     - inet_c     -   0
>> Jun28 -        00:00:00 -
>> 1 S 499          -     -  4511  0    -  99  19 -     - timerf     -   0
>> Jun28 -        00:00:18 -
>> 1 S 499          -     -  4512  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:00 -
>> 1 S 499          -     -  4513  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:00 -
>> 1 S 499          -     -  4536  0    -  99  19 -     - hrtime     -   0
>> Jun28 -        00:00:03 -
>> 1 S 499          -     -  4558  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:43 -
>> 1 S 499          -     -  4559  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:43 -
>> 1 S 499          -     -  4560  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:43 -
>> 1 S 499          -     -  4561  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:43 -
>> 1 S 499          -     -  4562  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:43 -
>> 1 S 499          -     -  4564  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:39 -
>> 1 S 499          -     -  4565  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:43 -
>> 1 S 499          -     -  4567  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:39 -
>> 1 S 499          -     -  4568  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:40 -
>> 1 S 499          -     -  4569  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:40 -
>> 1 S 499          -     -  4570  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:40 -
>> 1 S 499          -     -  4571  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:39 -
>> 1 S 499          -     -  4572  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:40 -
>> 1 S 499          -     -  4573  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:39 -
>> 1 S 499          -     -  4575  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:40 -
>> 1 S 499          -     -  4636  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:17 -
>> 1 S 499          -     -  4637  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:17 -
>> 1 S 499          -     -  4638  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:17 -
>> 1 S 499          -     -  4639  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:17 -
>> 1 S 499          -     -  4640  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:17 -
>> 1 S 499          -     -  4641  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:17 -
>> 1 S 499          -     -  4642  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:17 -
>> 1 S 499          -     -  4643  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:17 -
>> 1 S 499          -     -  4644  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:17 -
>> 1 S 499          -     -  4645  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:17 -
>> 1 S 499          -     -  4646  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:17 -
>> 1 S 499          -     -  4647  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:17 -
>> 1 S 499          -     -  4648  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:17 -
>> 1 S 499          -     -  4649  0    -  99  19 -     - futex_     -   0
>> Jun28 -        00:00:17 -
>>
>> step 2) *pstack 4481*
>>
>> Thread 45 (Thread 0x7ff6ba08a700 (LWP 4495)):
>> #0  0x0000003410ce8e63 in epoll_wait () from /lib64/libc.so.6
>> #1  0x00007ff6b877bba5 in ?? () from
>> /usr/local/freeswitch/mod/mod_sofia.so
>> #2  0x00007ff6b877ba28 in ?? () from
>> /usr/local/freeswitch/mod/mod_sofia.so
>> #3  0x00007ff6b8779715 in ?? () from
>> /usr/local/freeswitch/mod/mod_sofia.so
>> #4  0x00000034110079d1 in start_thread () from /lib64/libpthread.so.0
>> #5  0x0000003410ce886d in clone () from /lib64/libc.so.6
>>
>> Any idea?
>>
>>
>> Thanks,
>>
>> *PRESENCE TECHNOLOGY*
>> *Agust? Ubalde Bellot*
>> Chief Developer
>> C/ Comte Urgell 240 3A
>> Barcelona 08036
>> aubalde at presenceco.com
>>
>> Ph: +34 93 10 10 300
>> Fx: +34 93 10 10 333
>>
>> *www.presenceco.com* <http://www.presenceco.com/>
>>
>> *Follow us on:*
>>
>> *[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
>> <http://www.youtube.com/PresenceTechnology>*[image: in]*
>> <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]*
>> <http://www.slideshare.net/PresenceTech>*[image: fb]*
>> <http://www.facebook.com/presencetech>
>>
>> For additional information, please visit our website *www.presenceco.com*
>> <http://www.presenceco.com/>
>>
>>
>> *Presence Technology - DisclaimerThis message, its content and any file
>> attached thereto is for the intended recipient only and is confidential and
>> /or privileged. If you have received this e-mail in error or had access to
>> it, you should note that the information in it is private and any use
>> thereof is unauthorized. In such an event please notify us by e-mail or by
>> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by
>> whatsoever means and any transmission or dissemination thereof to other
>> persons is prohibited. It should be deleted immediately from your system.
>> Presence Technology reserves the right to take legal action against any
>> persons unlawfully gaining access to the content of any external message it
>> has emitted.*
>>
>> *For additional information, please visit our website **www.presenceco.com
>> <http://www.presenceco.com/>*
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 



*Presence Technology - DisclaimerThis message, its content and any file 
attached thereto is for the intended recipient only and is confidential and 
/or privileged. If you have received this e-mail in error or had access to 
it, you should note that the information in it is private and any use 
thereof is unauthorized. In such an event please notify us by e-mail or by 
telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by 
whatsoever means and any transmission or dissemination thereof to other 
persons is prohibited. It should be deleted immediately from your system. 
Presence Technology reserves the right to take legal action against any 
persons unlawfully gaining access to the content of any external message it 
has emitted.*

*For additional information, please visit our website **www.presenceco.com 
<http://www.presenceco.com/>*

 
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From gmaruzz at gmail.com  Wed Jun 29 12:53:36 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 29 Jun 2016 10:53:36 +0200
Subject: [Freeswitch-users] High CPU consumption
In-Reply-To: <CAKThY040oQGR8e-5+f2fsDC2ArG_EVXke8iX975A5tk=eqJVBw@mail.gmail.com>
References: <CAKThY07a4Jpnbbtv9C1kin09eZ9DJYRMXHmp34dK=kwOWBi5-A@mail.gmail.com>
	<CALXCt0r_bmCnmQqVyvTuvJ2yMDp=Hc_0hf86VmSw5u=mj_eaUg@mail.gmail.com>
	<CAKThY040oQGR8e-5+f2fsDC2ArG_EVXke8iX975A5tk=eqJVBw@mail.gmail.com>
Message-ID: <CALXCt0pqQOSxykF3g0kYa+_ym1jupGyWLXzOFxO0UCmXhc5NXw@mail.gmail.com>

I would counsel you to check on Debian 8 (Jessie), latest FreeSWITCH
release.

-giovanni

On Wed, Jun 29, 2016 at 10:37 AM, Agust? Ubalde <aubalde at presenceco.com>
wrote:

> Hi Giovanni,
>
> I don't use video muxing, only audio.
>
> This is a more detailed stack:
>
> Thread 45 (Thread 0x7ff6ba08a700 (LWP 4495)):
> #0  0x0000003410ce8e63 in epoll_wait () from /lib64/libc.so.6
> #1  0x00007ff6b877bba5 in su_epoll_port_wait_events () from
> /usr/local/freeswitch/mod/mod_sofia.so
> #2  0x00007ff6b877ba28 in su_base_port_run () from
> /usr/local/freeswitch/mod/mod_sofia.so
> #3  0x00007ff6b8779715 in su_pthread_port_clone_main () from
> /usr/local/freeswitch/mod/mod_sofia.so
> #4  0x00000034110079d1 in start_thread () from /lib64/libpthread.so.0
> #5  0x0000003410ce886d in clone () from /lib64/libc.so.6
>
>
> Thanks,
>
> *PRESENCE TECHNOLOGY*
> *Agust? Ubalde Bellot*
> Chief Developer
> C/ Comte Urgell 240 3A
> Barcelona 08036
> aubalde at presenceco.com
>
> Ph: +34 93 10 10 300
> Fx: +34 93 10 10 333
>
> *www.presenceco.com* <http://www.presenceco.com/>
>
> *Follow us on:*
>
> *[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
> <http://www.youtube.com/PresenceTechnology>*[image: in]*
> <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]*
> <http://www.slideshare.net/PresenceTech>*[image: fb]*
> <http://www.facebook.com/presencetech>
>
> For additional information, please visit our website *www.presenceco.com*
> <http://www.presenceco.com/>
>
> 2016-06-29 10:27 GMT+02:00 Giovanni Maruzzelli <gmaruzz at gmail.com>:
>
>> Video muxing is heavy on CPU, that's normal.
>>
>> Also, you will find best performances and reliability using Debian 8
>> Jessie, and latest FreeSWITCH release.
>>
>> Your specific version of FreeSWITCH is too old.
>>
>> CentOS 6 is not supposed to be used for video (webrtc) service.
>>
>> -giovanni
>>
>> On Wed, Jun 29, 2016 at 10:20 AM, Agust? Ubalde <aubalde at presenceco.com>
>> wrote:
>>
>>> Hi all,
>>>
>>> I have a high CPU consumption by FreeSWITCH. In fact, consumption is
>>> highest on CPU system. On this server we have registered 20 sessions
>>> (WebRTC).
>>>
>>> Environment:
>>>
>>>    - CentOS 6.7 64b
>>>    - FreeSWITCh 1.5.14
>>>
>>>
>>> The results of the execution of:
>>>
>>> step 1) *ps -LlFm -p `pidof freeswitch`*
>>>
>>> F S UID        PID  PPID   LWP  C NLWP PRI  NI ADDR SZ WCHAN    RSS PSR
>>> STIME TTY          TIME CMD
>>> 1 - 499       4481     1     - 99   55   -   - - 216700 -     140840  -
>>> Jun28 ?        17:55:10 /usr/bin/freeswitch -nc
>>> 1 S 499          -     -  4481  0    -  99  19 -     - poll_s     -   0
>>> Jun28 -        00:00:00 -
>>> 1 S 499          -     -  4482  0    -  99  19 -     - poll_s     -   0
>>> Jun28 -        00:00:00 -
>>> 1 S 499          -     -  4483  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:32 -
>>> 1 S 499          -     -  4484  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:03 -
>>> 1 S 499          -     -  4487  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:01 -
>>> 1 S 499          -     -  4488  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:04 -
>>> 1 S 499          -     -  4489  0    -  99  19 -     - poll_s     -   0
>>> Jun28 -        00:00:00 -
>>> 1 S 499          -     -  4490  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:01 -
>>> 1 S 499          -     -  4492  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:00 -
>>> 1 S 499          -     -  4493  0    -  99  19 -     - ep_pol     -   0
>>> Jun28 -        00:00:00 -
>>> 1 S 499          -     -  4494  0    -  99  19 -     - ep_pol     -   0
>>> Jun28 -        00:00:00 -
>>> *1 R 499          -     -  4495 97    -  99  19 -     - -          -   0
>>> Jun28 -        17:37:34 -*
>>> 1 S 499          -     -  4496  0    -  99  19 -     - ep_pol     -   0
>>> Jun28 -        00:00:00 -
>>> 1 S 499          -     -  4497  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:00 -
>>> 1 S 499          -     -  4498  0    -  99  19 -     - poll_s     -   0
>>> Jun28 -        00:00:00 -
>>> 1 S 499          -     -  4499  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:00 -
>>> 1 S 499          -     -  4500  0    -  99  19 -     - poll_s     -   0
>>> Jun28 -        00:00:01 -
>>> 1 S 499          -     -  4501  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:00 -
>>> 1 S 499          -     -  4502  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:00 -
>>> 1 S 499          -     -  4503  0    -  99  19 -     - poll_s     -   0
>>> Jun28 -        00:00:00 -
>>> 1 S 499          -     -  4504  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:00 -
>>> 1 S 499          -     -  4510  0    -  99  19 -     - inet_c     -   0
>>> Jun28 -        00:00:00 -
>>> 1 S 499          -     -  4511  0    -  99  19 -     - timerf     -   0
>>> Jun28 -        00:00:18 -
>>> 1 S 499          -     -  4512  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:00 -
>>> 1 S 499          -     -  4513  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:00 -
>>> 1 S 499          -     -  4536  0    -  99  19 -     - hrtime     -   0
>>> Jun28 -        00:00:03 -
>>> 1 S 499          -     -  4558  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:43 -
>>> 1 S 499          -     -  4559  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:43 -
>>> 1 S 499          -     -  4560  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:43 -
>>> 1 S 499          -     -  4561  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:43 -
>>> 1 S 499          -     -  4562  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:43 -
>>> 1 S 499          -     -  4564  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:39 -
>>> 1 S 499          -     -  4565  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:43 -
>>> 1 S 499          -     -  4567  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:39 -
>>> 1 S 499          -     -  4568  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:40 -
>>> 1 S 499          -     -  4569  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:40 -
>>> 1 S 499          -     -  4570  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:40 -
>>> 1 S 499          -     -  4571  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:39 -
>>> 1 S 499          -     -  4572  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:40 -
>>> 1 S 499          -     -  4573  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:39 -
>>> 1 S 499          -     -  4575  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:40 -
>>> 1 S 499          -     -  4636  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:17 -
>>> 1 S 499          -     -  4637  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:17 -
>>> 1 S 499          -     -  4638  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:17 -
>>> 1 S 499          -     -  4639  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:17 -
>>> 1 S 499          -     -  4640  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:17 -
>>> 1 S 499          -     -  4641  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:17 -
>>> 1 S 499          -     -  4642  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:17 -
>>> 1 S 499          -     -  4643  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:17 -
>>> 1 S 499          -     -  4644  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:17 -
>>> 1 S 499          -     -  4645  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:17 -
>>> 1 S 499          -     -  4646  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:17 -
>>> 1 S 499          -     -  4647  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:17 -
>>> 1 S 499          -     -  4648  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:17 -
>>> 1 S 499          -     -  4649  0    -  99  19 -     - futex_     -   0
>>> Jun28 -        00:00:17 -
>>>
>>> step 2) *pstack 4481*
>>>
>>> Thread 45 (Thread 0x7ff6ba08a700 (LWP 4495)):
>>> #0  0x0000003410ce8e63 in epoll_wait () from /lib64/libc.so.6
>>> #1  0x00007ff6b877bba5 in ?? () from
>>> /usr/local/freeswitch/mod/mod_sofia.so
>>> #2  0x00007ff6b877ba28 in ?? () from
>>> /usr/local/freeswitch/mod/mod_sofia.so
>>> #3  0x00007ff6b8779715 in ?? () from
>>> /usr/local/freeswitch/mod/mod_sofia.so
>>> #4  0x00000034110079d1 in start_thread () from /lib64/libpthread.so.0
>>> #5  0x0000003410ce886d in clone () from /lib64/libc.so.6
>>>
>>> Any idea?
>>>
>>>
>>> Thanks,
>>>
>>> *PRESENCE TECHNOLOGY*
>>> *Agust? Ubalde Bellot*
>>> Chief Developer
>>> C/ Comte Urgell 240 3A
>>> Barcelona 08036
>>> aubalde at presenceco.com
>>>
>>> Ph: +34 93 10 10 300
>>> Fx: +34 93 10 10 333
>>>
>>> *www.presenceco.com* <http://www.presenceco.com/>
>>>
>>> *Follow us on:*
>>>
>>> *[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
>>> <http://www.youtube.com/PresenceTechnology>*[image: in]*
>>> <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]*
>>> <http://www.slideshare.net/PresenceTech>*[image: fb]*
>>> <http://www.facebook.com/presencetech>
>>>
>>> For additional information, please visit our website
>>> *www.presenceco.com* <http://www.presenceco.com/>
>>>
>>>
>>> *Presence Technology - DisclaimerThis message, its content and any file
>>> attached thereto is for the intended recipient only and is confidential and
>>> /or privileged. If you have received this e-mail in error or had access to
>>> it, you should note that the information in it is private and any use
>>> thereof is unauthorized. In such an event please notify us by e-mail or by
>>> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by
>>> whatsoever means and any transmission or dissemination thereof to other
>>> persons is prohibited. It should be deleted immediately from your system.
>>> Presence Technology reserves the right to take legal action against any
>>> persons unlawfully gaining access to the content of any external message it
>>> has emitted.*
>>>
>>> *For additional information, please visit our website **www.presenceco.com
>>> <http://www.presenceco.com/>*
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> *Presence Technology - DisclaimerThis message, its content and any file
> attached thereto is for the intended recipient only and is confidential and
> /or privileged. If you have received this e-mail in error or had access to
> it, you should note that the information in it is private and any use
> thereof is unauthorized. In such an event please notify us by e-mail or by
> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by
> whatsoever means and any transmission or dissemination thereof to other
> persons is prohibited. It should be deleted immediately from your system.
> Presence Technology reserves the right to take legal action against any
> persons unlawfully gaining access to the content of any external message it
> has emitted.*
>
> *For additional information, please visit our website **www.presenceco.com
> <http://www.presenceco.com/>*
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From jurijs.ivolga at gmail.com  Wed Jun 29 17:21:28 2016
From: jurijs.ivolga at gmail.com (Jurijs Ivolga)
Date: Wed, 29 Jun 2016 16:21:28 +0300
Subject: [Freeswitch-users] Video Refresh INFO requests
Message-ID: <CAOrA2UYbKM94+xmBP17-Yw=ek+Q5KgGyMOMYxCKRgWM05er81w@mail.gmail.com>

Hi,

My Freeswitch server is flooding with video refresh INFO method like this:

INFO sip:aaaa at x.x.x.x:61838;transport=tls SIP/2.0.
> Via: SIP/2.0/UDP x.x.x.x;rport;branch=z9hG4bKKr3U2HQyU08yc.
> Route: <sip:x.x.x.x;lr>.
> Route: <sip:x.x.x.x;r2=on;lr>.
> Route: <sip:x.x.x.x:5061;transport=tls;r2=on;lr>.
> Max-Forwards: 70.
> From: "aaaa" <sip:aaaa at voip.net>;tag=3UcX6HB3Q98ZK.
> To: <sip:bbb at voip.net>;tag=626GIwP.
> Call-ID: 0f55526d-b886-1234-e784-040185d50001.
> CSeq: 93270262 INFO.
> Contact: <sip:mod_sofia at x.x.x.x:5060>.
> User-Agent:
> FreeSWITCH-mod_sofia/1.6.7+git~20160401T134007Z~f0c3870be3~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Content-Type: application/media_control+xml.
> Content-Length: 149.
> .
> <?xml version="1.0" encoding="utf-8" ?>
> <media_control><vc_primitive><to_encoder><picture_fast_update
> /></to_encoder></vc_primitive></media_control>
>

Is it possible somehow to switch this off?

Jurijs
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From anthony.minessale at gmail.com  Wed Jun 29 19:43:47 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 29 Jun 2016 10:43:47 -0500
Subject: [Freeswitch-users] Video Refresh INFO requests
In-Reply-To: <CAOrA2UYbKM94+xmBP17-Yw=ek+Q5KgGyMOMYxCKRgWM05er81w@mail.gmail.com>
References: <CAOrA2UYbKM94+xmBP17-Yw=ek+Q5KgGyMOMYxCKRgWM05er81w@mail.gmail.com>
Message-ID: <CAKbxfG92Q1v70qy3PuiE8nXDbcAGTAJNaomTbrHCNV-mFRRHMA@mail.gmail.com>

not currently but if you file a jira we can probably add a profile param to
disable it

On Wed, Jun 29, 2016 at 8:21 AM, Jurijs Ivolga <jurijs.ivolga at gmail.com>
wrote:

> Hi,
>
> My Freeswitch server is flooding with video refresh INFO method like this:
>
> INFO sip:aaaa at x.x.x.x:61838;transport=tls SIP/2.0.
>> Via: SIP/2.0/UDP x.x.x.x;rport;branch=z9hG4bKKr3U2HQyU08yc.
>> Route: <sip:x.x.x.x;lr>.
>> Route: <sip:x.x.x.x;r2=on;lr>.
>> Route: <sip:x.x.x.x:5061;transport=tls;r2=on;lr>.
>> Max-Forwards: 70.
>> From: "aaaa" <sip:aaaa at voip.net>;tag=3UcX6HB3Q98ZK.
>> To: <sip:bbb at voip.net>;tag=626GIwP.
>> Call-ID: 0f55526d-b886-1234-e784-040185d50001.
>> CSeq: 93270262 INFO.
>> Contact: <sip:mod_sofia at x.x.x.x:5060>.
>> User-Agent:
>> FreeSWITCH-mod_sofia/1.6.7+git~20160401T134007Z~f0c3870be3~64bit.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY.
>> Supported: timer, path, replaces.
>> Content-Type: application/media_control+xml.
>> Content-Length: 149.
>> .
>> <?xml version="1.0" encoding="utf-8" ?>
>> <media_control><vc_primitive><to_encoder><picture_fast_update
>> /></to_encoder></vc_primitive></media_control>
>>
>
> Is it possible somehow to switch this off?
>
> Jurijs
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From mandra at gmail.com  Wed Jun 29 21:44:44 2016
From: mandra at gmail.com (Chris Mandra)
Date: Wed, 29 Jun 2016 13:44:44 -0400
Subject: [Freeswitch-users] Codec Reset Issue With Media Bug Invoked On
 Both Call Legs
In-Reply-To: <94837390-55C1-49CF-8FEF-B8D288AE2914@gmail.com>
References: <021848EC-E397-4EF7-8073-E7CD6F2C9EFA@gmail.com>
	<CAKbxfG_=mDhhbnEMPQSFi4pirQkAdJ6T8tjPC48ZNB2=YiVh4Q@mail.gmail.com>
	<94837390-55C1-49CF-8FEF-B8D288AE2914@gmail.com>
Message-ID: <CAJdRzN6gig0Jpj6_mkBvowfUhkEio40m4B+oX5iBCz68VDr+sQ@mail.gmail.com>

I'm curious about this too -any thoughts?
On Fri, Jun 24, 2016 at 6:45 PM, Jon Lederman <jonlederman at gmail.com> wrote:

> Hi,
>
>
> By codec reset I mean the following:
>
> 2016-05-04 03:51:32.464652 [DEBUG] mod_opus.c:661 Opus decoder stats:
> Frames[0] PLC[0] FEC[0]
> 2016-05-04 03:51:32.464652 [DEBUG] mod_opus.c:559 Opus encoder: set
> bitrate to local settings [72000bps]
>
> Thanks
>
> Jon
>
>
> Sent from my iPhone
>
> On Jun 24, 2016, at 6:34 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
> What is a codec reset?
>
> On Friday, June 24, 2016, Jon Lederman <jonlederman at gmail.com> wrote:
>
>> Hi,
>>
>> We are utilizing a module that performs processing on audio using a media
>> bug.  We are finding that if the media bug is invoked on only one call leg,
>> we are seeing very frequent codec resets on the call leg that doesn?t have
>> the media bug invoked leading to very choppy audio.  If the media bug is on
>> both call legs we don?t see the issue or the number of codec resets is much
>> lower.
>>
>> Any thoughts on what may be causing this and why having the media bug on
>> both call legs would make a difference?
>>
>>
>> Thanks.
>>
>> -Jon
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
mandra
c:410.258.5281
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From anthony.minessale at gmail.com  Thu Jun 30 00:21:37 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 29 Jun 2016 15:21:37 -0500
Subject: [Freeswitch-users] Codec Reset Issue With Media Bug Invoked On
 Both Call Legs
In-Reply-To: <CAJdRzN6gig0Jpj6_mkBvowfUhkEio40m4B+oX5iBCz68VDr+sQ@mail.gmail.com>
References: <021848EC-E397-4EF7-8073-E7CD6F2C9EFA@gmail.com>
	<CAKbxfG_=mDhhbnEMPQSFi4pirQkAdJ6T8tjPC48ZNB2=YiVh4Q@mail.gmail.com>
	<94837390-55C1-49CF-8FEF-B8D288AE2914@gmail.com>
	<CAJdRzN6gig0Jpj6_mkBvowfUhkEio40m4B+oX5iBCz68VDr+sQ@mail.gmail.com>
Message-ID: <CAKbxfG90iNt_Tj5VbxkfRd05Yv6RBdyKDkYUvFLh2g_7A=UC4g@mail.gmail.com>

you could try

uuid_codec_debug <uuid> audio 10


and look for some hints.



On Wed, Jun 29, 2016 at 12:44 PM, Chris Mandra <mandra at gmail.com> wrote:

> I'm curious about this too -any thoughts?
> On Fri, Jun 24, 2016 at 6:45 PM, Jon Lederman <jonlederman at gmail.com>
> wrote:
>
>> Hi,
>>
>>
>> By codec reset I mean the following:
>>
>> 2016-05-04 03:51:32.464652 [DEBUG] mod_opus.c:661 Opus decoder stats:
>> Frames[0] PLC[0] FEC[0]
>> 2016-05-04 03:51:32.464652 [DEBUG] mod_opus.c:559 Opus encoder: set
>> bitrate to local settings [72000bps]
>>
>> Thanks
>>
>> Jon
>>
>>
>> Sent from my iPhone
>>
>> On Jun 24, 2016, at 6:34 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>> What is a codec reset?
>>
>> On Friday, June 24, 2016, Jon Lederman <jonlederman at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> We are utilizing a module that performs processing on audio using a
>>> media bug.  We are finding that if the media bug is invoked on only one
>>> call leg, we are seeing very frequent codec resets on the call leg that
>>> doesn?t have the media bug invoked leading to very choppy audio.  If the
>>> media bug is on both call legs we don?t see the issue or the number of
>>> codec resets is much lower.
>>>
>>> Any thoughts on what may be causing this and why having the media bug on
>>> both call legs would make a difference?
>>>
>>>
>>> Thanks.
>>>
>>> -Jon
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>>
>> ? http://freeswitch.org/  ? http://cluecon.com/  ?
>> http://twitter.com/FreeSWITCH
>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
>> <http://freeswitch.org/g+>*
>>
>> ClueCon Weekly Development Call
>> ? sip:888 at conference.freeswitch.org  ? +19193869900
>>
>> https://www.youtube.com/watch?v=9XXgW34t40s
>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> mandra
> c:410.258.5281
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From gregor at infomedia.si  Thu Jun 30 01:26:23 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Wed, 29 Jun 2016 23:26:23 +0200
Subject: [Freeswitch-users] Verto + 180 Ringing
Message-ID: <CAC7ttFHAvAht8pozarB0d6qt4PoiZVYL_3x48wTuxUY7QeuEew@mail.gmail.com>

Hi!

Can someone plese help me.

I have strange behaviour on outbound calls. Provider that terminates calls
sends only SIP 180 , not also SIP183. And it looks that verto doesn't
forward 180 to client and there is no ringing tone in verto client neither
I can capture this event in client. I tried to ring_ready, but verto still
doesn't send anything to client (originator). It only sends event on 183.

Is this by design? How can I set in dialplan and send client 183?

Thank you, Gregor
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From anthony.minessale at gmail.com  Thu Jun 30 01:42:13 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 29 Jun 2016 16:42:13 -0500
Subject: [Freeswitch-users] Verto + 180 Ringing
In-Reply-To: <CAC7ttFHAvAht8pozarB0d6qt4PoiZVYL_3x48wTuxUY7QeuEew@mail.gmail.com>
References: <CAC7ttFHAvAht8pozarB0d6qt4PoiZVYL_3x48wTuxUY7QeuEew@mail.gmail.com>
Message-ID: <CAKbxfG8_-b7uRDK_gykgKxkszrA3jkKuq0eDj3VyJPWcT6PGEg@mail.gmail.com>

Try setting the ringback variable to an appropriate tone ${us-ring} and
${uk-ring} are pre-defined I think.


On Wed, Jun 29, 2016 at 4:26 PM, Gregor Nanger <gregor at infomedia.si> wrote:

> Hi!
>
> Can someone plese help me.
>
> I have strange behaviour on outbound calls. Provider that terminates calls
> sends only SIP 180 , not also SIP183. And it looks that verto doesn't
> forward 180 to client and there is no ringing tone in verto client neither
> I can capture this event in client. I tried to ring_ready, but verto still
> doesn't send anything to client (originator). It only sends event on 183.
>
> Is this by design? How can I set in dialplan and send client 183?
>
> Thank you, Gregor
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
-------------- next part --------------
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From gregor at infomedia.si  Thu Jun 30 02:07:17 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Thu, 30 Jun 2016 00:07:17 +0200
Subject: [Freeswitch-users] Verto + 180 Ringing
In-Reply-To: <CAKbxfG8_-b7uRDK_gykgKxkszrA3jkKuq0eDj3VyJPWcT6PGEg@mail.gmail.com>
References: <CAC7ttFHAvAht8pozarB0d6qt4PoiZVYL_3x48wTuxUY7QeuEew@mail.gmail.com>
	<CAKbxfG8_-b7uRDK_gykgKxkszrA3jkKuq0eDj3VyJPWcT6PGEg@mail.gmail.com>
Message-ID: <CAC7ttFEwk-q0UJm6Ptaz_j+Oji8uBmUmwTmvn82ChPTK_P=ngw@mail.gmail.com>

Thank you, Anthony. This does the trick.

Side effect is that on busy it still rings once,twice, but it is way better
than no ringing.


2016-06-29 23:42 GMT+02:00 Anthony Minessale <anthony.minessale at gmail.com>:

> Try setting the ringback variable to an appropriate tone ${us-ring} and
> ${uk-ring} are pre-defined I think.
>
>
> On Wed, Jun 29, 2016 at 4:26 PM, Gregor Nanger <gregor at infomedia.si>
> wrote:
>
>> Hi!
>>
>> Can someone plese help me.
>>
>> I have strange behaviour on outbound calls. Provider that terminates
>> calls sends only SIP 180 , not also SIP183. And it looks that verto doesn't
>> forward 180 to client and there is no ringing tone in verto client neither
>> I can capture this event in client. I tried to ring_ready, but verto still
>> doesn't send anything to client (originator). It only sends event on 183.
>>
>> Is this by design? How can I set in dialplan and send client 183?
>>
>> Thank you, Gregor
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
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
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From mike at jerris.com  Thu Jun 30 02:13:55 2016
From: mike at jerris.com (Michael Jerris)
Date: Wed, 29 Jun 2016 18:13:55 -0400
Subject: [Freeswitch-users] Verto + 180 Ringing
In-Reply-To: <CAC7ttFEwk-q0UJm6Ptaz_j+Oji8uBmUmwTmvn82ChPTK_P=ngw@mail.gmail.com>
References: <CAC7ttFHAvAht8pozarB0d6qt4PoiZVYL_3x48wTuxUY7QeuEew@mail.gmail.com>
	<CAKbxfG8_-b7uRDK_gykgKxkszrA3jkKuq0eDj3VyJPWcT6PGEg@mail.gmail.com>
	<CAC7ttFEwk-q0UJm6Ptaz_j+Oji8uBmUmwTmvn82ChPTK_P=ngw@mail.gmail.com>
Message-ID: <5FDEDEC2-265F-4466-BB32-008C17C581AF@jerris.com>

What is the sip message flow that leads to ring then busy?  Are you actually getting a 180 before the busy?

> On Jun 29, 2016, at 6:07 PM, Gregor Nanger <gregor at infomedia.si> wrote:
> 
> Thank you, Anthony. This does the trick. 
> 
> Side effect is that on busy it still rings once,twice, but it is way better than no ringing.
> 
> 
> 2016-06-29 23:42 GMT+02:00 Anthony Minessale <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>>:
> Try setting the ringback variable to an appropriate tone ${us-ring} and ${uk-ring} are pre-defined I think.
> 
> 
> On Wed, Jun 29, 2016 at 4:26 PM, Gregor Nanger <gregor at infomedia.si <mailto:gregor at infomedia.si>> wrote:
> Hi!
> 
> Can someone plese help me.
> 
> I have strange behaviour on outbound calls. Provider that terminates calls sends only SIP 180 , not also SIP183. And it looks that verto doesn't forward 180 to client and there is no ringing tone in verto client neither I can capture this event in client. I tried to ring_ready, but verto still doesn't send anything to client (originator). It only sends event on 183.
> 
> Is this by design? How can I set in dialplan and send client 183?
> 
> Thank you, Gregor
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
> 
> ? http://freeswitch.org/ <http://freeswitch.org/>  ? http://cluecon.com/ <http://cluecon.com/>  ? http://twitter.com/FreeSWITCH <http://twitter.com/FreeSWITCH>
> ? irc.freenode.net <http://irc.freenode.net/> #freeswitch ? http://freeswitch.org/g+ <http://freeswitch.org/g+>
> 
> ClueCon Weekly Development Call 
> ? sip:888 at conference.freeswitch.org <mailto:sip%3A888 at conference.freeswitch.org>  ? +19193869900 <tel:%2B19193869900> 
> 
> https://www.youtube.com/watch?v=9XXgW34t40s <https://www.youtube.com/watch?v=9XXgW34t40s>
> https://www.youtube.com/watch?v=NLaDpGQuZDA <https://www.youtube.com/watch?v=NLaDpGQuZDA>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> 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/>_________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From gregor at infomedia.si  Thu Jun 30 02:30:19 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Thu, 30 Jun 2016 00:30:19 +0200
Subject: [Freeswitch-users] Verto + 180 Ringing
In-Reply-To: <5FDEDEC2-265F-4466-BB32-008C17C581AF@jerris.com>
References: <CAC7ttFHAvAht8pozarB0d6qt4PoiZVYL_3x48wTuxUY7QeuEew@mail.gmail.com>
	<CAKbxfG8_-b7uRDK_gykgKxkszrA3jkKuq0eDj3VyJPWcT6PGEg@mail.gmail.com>
	<CAC7ttFEwk-q0UJm6Ptaz_j+Oji8uBmUmwTmvn82ChPTK_P=ngw@mail.gmail.com>
	<5FDEDEC2-265F-4466-BB32-008C17C581AF@jerris.com>
Message-ID: <CAC7ttFGTiqiAjOu6ydHcPud0wxUxcM0Vg4xQOQcfiK9ZxGpi6Q@mail.gmail.com>

Hmm.... I didn't pay attention, but now I see that it is funny how get flow
from provider when callee is busy:

INVITE
SIP 100
SIP 180
SIP 488

guess they send ringing anyway...

2016-06-30 0:13 GMT+02:00 Michael Jerris <mike at jerris.com>:

> What is the sip message flow that leads to ring then busy?  Are you
> actually getting a 180 before the busy?
>
> On Jun 29, 2016, at 6:07 PM, Gregor Nanger <gregor at infomedia.si> wrote:
>
> Thank you, Anthony. This does the trick.
>
> Side effect is that on busy it still rings once,twice, but it is way
> better than no ringing.
>
>
> 2016-06-29 23:42 GMT+02:00 Anthony Minessale <anthony.minessale at gmail.com>
> :
>
>> Try setting the ringback variable to an appropriate tone ${us-ring} and
>> ${uk-ring} are pre-defined I think.
>>
>>
>> On Wed, Jun 29, 2016 at 4:26 PM, Gregor Nanger <gregor at infomedia.si>
>> wrote:
>>
>>> Hi!
>>>
>>> Can someone plese help me.
>>>
>>> I have strange behaviour on outbound calls. Provider that terminates
>>> calls sends only SIP 180 , not also SIP183. And it looks that verto doesn't
>>> forward 180 to client and there is no ringing tone in verto client neither
>>> I can capture this event in client. I tried to ring_ready, but verto still
>>> doesn't send anything to client (originator). It only sends event on 183.
>>>
>>> Is this by design? How can I set in dialplan and send client 183?
>>>
>>> Thank you, Gregor
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>>
>> ? http://freeswitch.org/  ? http://cluecon.com/  ?
>> http://twitter.com/FreeSWITCH
>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
>> <http://freeswitch.org/g+>*
>>
>> ClueCon Weekly Development Call
>> ? sip:888 at conference.freeswitch.org  ? +19193869900
>>
>> https://www.youtube.com/watch?v=9XXgW34t40s
>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> 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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
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
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From mike at jerris.com  Thu Jun 30 02:46:16 2016
From: mike at jerris.com (Michael Jerris)
Date: Wed, 29 Jun 2016 18:46:16 -0400
Subject: [Freeswitch-users] Verto + 180 Ringing
In-Reply-To: <CAC7ttFGTiqiAjOu6ydHcPud0wxUxcM0Vg4xQOQcfiK9ZxGpi6Q@mail.gmail.com>
References: <CAC7ttFHAvAht8pozarB0d6qt4PoiZVYL_3x48wTuxUY7QeuEew@mail.gmail.com>
	<CAKbxfG8_-b7uRDK_gykgKxkszrA3jkKuq0eDj3VyJPWcT6PGEg@mail.gmail.com>
	<CAC7ttFEwk-q0UJm6Ptaz_j+Oji8uBmUmwTmvn82ChPTK_P=ngw@mail.gmail.com>
	<5FDEDEC2-265F-4466-BB32-008C17C581AF@jerris.com>
	<CAC7ttFGTiqiAjOu6ydHcPud0wxUxcM0Vg4xQOQcfiK9ZxGpi6Q@mail.gmail.com>
Message-ID: <F3A0E1D6-92C2-41CF-90E1-98791510F60C@jerris.com>

So provider is actually sending ring then busy.  Thats quite wrong.. You should make them fix that.


> On Jun 29, 2016, at 6:30 PM, Gregor Nanger <gregor at infomedia.si> wrote:
> 
> Hmm.... I didn't pay attention, but now I see that it is funny how get flow from provider when callee is busy:
> 
> INVITE
> SIP 100
> SIP 180
> SIP 488
> 
> guess they send ringing anyway...
> 
> 2016-06-30 0:13 GMT+02:00 Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>>:
> What is the sip message flow that leads to ring then busy?  Are you actually getting a 180 before the busy?
> 
>> On Jun 29, 2016, at 6:07 PM, Gregor Nanger <gregor at infomedia.si <mailto:gregor at infomedia.si>> wrote:
>> 
>> Thank you, Anthony. This does the trick. 
>> 
>> Side effect is that on busy it still rings once,twice, but it is way better than no ringing.
>> 
>> 
>> 2016-06-29 23:42 GMT+02:00 Anthony Minessale <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>>:
>> Try setting the ringback variable to an appropriate tone ${us-ring} and ${uk-ring} are pre-defined I think.
>> 
>> 
>> On Wed, Jun 29, 2016 at 4:26 PM, Gregor Nanger <gregor at infomedia.si <mailto:gregor at infomedia.si>> wrote:
>> Hi!
>> 
>> Can someone plese help me.
>> 
>> I have strange behaviour on outbound calls. Provider that terminates calls sends only SIP 180 , not also SIP183. And it looks that verto doesn't forward 180 to client and there is no ringing tone in verto client neither I can capture this event in client. I tried to ring_ready, but verto still doesn't send anything to client (originator). It only sends event on 183.
>> 
>> Is this by design? How can I set in dialplan and send client 183?
>> 
>> Thank you, Gregor
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> 
>> 
>> 
>> -- 
>> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>> 
>> ? http://freeswitch.org/ <http://freeswitch.org/>  ? http://cluecon.com/ <http://cluecon.com/>  ? http://twitter.com/FreeSWITCH <http://twitter.com/FreeSWITCH>
>> ? irc.freenode.net <http://irc.freenode.net/> #freeswitch ? http://freeswitch.org/g+ <http://freeswitch.org/g+>
>> 
>> ClueCon Weekly Development Call 
>> ? sip:888 at conference.freeswitch.org <mailto:sip%3A888 at conference.freeswitch.org>  ? +19193869900 <tel:%2B19193869900> 
>> 
>> https://www.youtube.com/watch?v=9XXgW34t40s <https://www.youtube.com/watch?v=9XXgW34t40s>
>> https://www.youtube.com/watch?v=NLaDpGQuZDA <https://www.youtube.com/watch?v=NLaDpGQuZDA>
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> 
>> 
>> 
>> -- 
>> 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/>_________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: 
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> 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/>_________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From gregor at infomedia.si  Thu Jun 30 02:53:04 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Thu, 30 Jun 2016 00:53:04 +0200
Subject: [Freeswitch-users] Verto + 180 Ringing
In-Reply-To: <F3A0E1D6-92C2-41CF-90E1-98791510F60C@jerris.com>
References: <CAC7ttFHAvAht8pozarB0d6qt4PoiZVYL_3x48wTuxUY7QeuEew@mail.gmail.com>
	<CAKbxfG8_-b7uRDK_gykgKxkszrA3jkKuq0eDj3VyJPWcT6PGEg@mail.gmail.com>
	<CAC7ttFEwk-q0UJm6Ptaz_j+Oji8uBmUmwTmvn82ChPTK_P=ngw@mail.gmail.com>
	<5FDEDEC2-265F-4466-BB32-008C17C581AF@jerris.com>
	<CAC7ttFGTiqiAjOu6ydHcPud0wxUxcM0Vg4xQOQcfiK9ZxGpi6Q@mail.gmail.com>
	<F3A0E1D6-92C2-41CF-90E1-98791510F60C@jerris.com>
Message-ID: <CAC7ttFFBZKjAFX3_EpNM22=6U-OuxF4oun7+m-UCzZFUObAbqw@mail.gmail.com>

Yes, I see. Thank you very much for your help...

2016-06-30 0:46 GMT+02:00 Michael Jerris <mike at jerris.com>:

> So provider is actually sending ring then busy.  Thats quite wrong.. You
> should make them fix that.
>
>
> On Jun 29, 2016, at 6:30 PM, Gregor Nanger <gregor at infomedia.si> wrote:
>
> Hmm.... I didn't pay attention, but now I see that it is funny how get
> flow from provider when callee is busy:
>
> INVITE
> SIP 100
> SIP 180
> SIP 488
>
> guess they send ringing anyway...
>
> 2016-06-30 0:13 GMT+02:00 Michael Jerris <mike at jerris.com>:
>
>> What is the sip message flow that leads to ring then busy?  Are you
>> actually getting a 180 before the busy?
>>
>> On Jun 29, 2016, at 6:07 PM, Gregor Nanger <gregor at infomedia.si> wrote:
>>
>> Thank you, Anthony. This does the trick.
>>
>> Side effect is that on busy it still rings once,twice, but it is way
>> better than no ringing.
>>
>>
>> 2016-06-29 23:42 GMT+02:00 Anthony Minessale <anthony.minessale at gmail.com
>> >:
>>
>>> Try setting the ringback variable to an appropriate tone ${us-ring} and
>>> ${uk-ring} are pre-defined I think.
>>>
>>>
>>> On Wed, Jun 29, 2016 at 4:26 PM, Gregor Nanger <gregor at infomedia.si>
>>> wrote:
>>>
>>>> Hi!
>>>>
>>>> Can someone plese help me.
>>>>
>>>> I have strange behaviour on outbound calls. Provider that terminates
>>>> calls sends only SIP 180 , not also SIP183. And it looks that verto doesn't
>>>> forward 180 to client and there is no ringing tone in verto client neither
>>>> I can capture this event in client. I tried to ring_ready, but verto still
>>>> doesn't send anything to client (originator). It only sends event on 183.
>>>>
>>>> Is this by design? How can I set in dialplan and send client 183?
>>>>
>>>> Thank you, Gregor
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>>>
>>> ? http://freeswitch.org/  ? http://cluecon.com/  ?
>>> http://twitter.com/FreeSWITCH
>>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
>>> <http://freeswitch.org/g+>*
>>>
>>> ClueCon Weekly Development Call
>>> ? sip:888 at conference.freeswitch.org  ? +19193869900
>>>
>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> 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
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> 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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
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
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From royj at yandex.ru  Thu Jun 30 11:16:42 2016
From: royj at yandex.ru (royj at yandex.ru)
Date: Thu, 30 Jun 2016 10:16:42 +0300
Subject: [Freeswitch-users] parameters in the Contact
Message-ID: <1888391467271002@web7j.yandex.ru>

Hi
We have an UA that sends in REGISTER to FreeSWITCH in Contact header some parameter. And that UA expects that paramter in INVITE RURI FreeSWITCH sends to it.
Is there ability to convey parameter in INVITE?


From royj at yandex.ru  Thu Jun 30 13:28:05 2016
From: royj at yandex.ru (royj at yandex.ru)
Date: Thu, 30 Jun 2016 12:28:05 +0300
Subject: [Freeswitch-users] parameters in the Contact
In-Reply-To: <1888391467271002@web7j.yandex.ru>
References: <1888391467271002@web7j.yandex.ru>
Message-ID: <555301467278885@web22m.yandex.ru>

Thank for attention, my mistake

30.06.2016, 10:21, "royj at yandex.ru" <royj at yandex.ru>:
> Hi
> We have an UA that sends in REGISTER to FreeSWITCH in Contact header some parameter. And that UA expects that paramter in INVITE RURI FreeSWITCH sends to it.
> Is there ability to convey parameter in INVITE?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From rick.jarvis at magicmail.mooo.com  Thu Jun 30 16:49:52 2016
From: rick.jarvis at magicmail.mooo.com (Rick Jarvis)
Date: Thu, 30 Jun 2016 13:49:52 +0100
Subject: [Freeswitch-users] Security vs compatibility / NAT etc
Message-ID: <C39782AE-E274-4417-A186-B042688E86EB@magicmail.mooo.com>

I?d be interested to hear what different people use to provide some level of security for remote end-users such as homeworkers, and to get round NAT issues.

We currently use OpenVPN, as this is built into the firmware of Yealink handsets (it?s a great feature, I?m not sure why more handset manufacturers don?t do this?!). The pros are that not only is it secure, but it also removes any problems with NAT for the RTP streams.

The downsides are that it is complicated (and downright frustrating sometimes) to set up, and there are additional things to consider such as the server configuration and overheads.

TLS/SSL with SRTP is another option, but my understanding of this is that it can cause NAT problems, with FreeSWITCH trying to initiate control channels back to the phone for inbound calls. In fact, I?ve always had problems with getting phones to work when behind NAT anyway, even without SSL/TLS. STUN can be used to ascertain the IP, but how do you handle situations where multiple handsets are behind NAT - you can?t open all RTP ports to all handsets at once?!!

Would be very interested to hear thoughts and methods on these points.

Thanks
R

From colin.morelli at gmail.com  Thu Jun 30 17:39:42 2016
From: colin.morelli at gmail.com (Colin Morelli)
Date: Thu, 30 Jun 2016 13:39:42 +0000
Subject: [Freeswitch-users] Security vs compatibility / NAT etc
In-Reply-To: <C39782AE-E274-4417-A186-B042688E86EB@magicmail.mooo.com>
References: <C39782AE-E274-4417-A186-B042688E86EB@magicmail.mooo.com>
Message-ID: <CAPtU-UqLTYJjUj6iKQkUPt5TqFq5dWqxoq4NhVqLVi7ytOESKA@mail.gmail.com>

Rick,

(Sorry for the long email, hopefully it's helpful)

It sounds like you're mostly concerned with FS initiating calls to handsets
behind NAT, is that correct?

If so, what you probably want is SIP outbound (RFC 5626). It's the best way
to avoid NAT issues with clients. Under this model, clients keep a
persistent connection open to the server. The server is responsible for
using that connection to deliver INVITEs to the client, thus avoiding the
need to ever open its own connection.

In my (relatively limited) experience with FS, it was able to act like a
SIP outbound server, but it doesn't directly advertise it and supporting
SIP outbound is really outside of the core scope of what FS does. So, in my
setup, I use Kamailio to provide the SIP outbound support. A brief
description of my setup (which seems to work fine with clients behind NAT)

Kamailio edge proxy cluster (provides SIP outbound support to clients,
allows public SIP traffic)
Kamailio proxy + registrar (only allows SIP traffic from inside the local
network, provides registration support)
Freeswitch (only allows SIP traffic from inside the local network, has a
public IP address and open firewall for RTP traffic).

So, a registration from a client hits the Kamailio edge proxy, which parks
the socket connection and sends it on to the second Kamailio
proxy/registrar. When FS needs to make outbound calls to clients, it hits
the Kamailio proxy/registrar, which forwards it to the edge proxy that has
an existing connection the client and uses it to deliver the invite (this
is all handled by Kamailio with it's outbound, path, registrar, and usrloc
modules).

Note your setup might not require the use of two layers of proxies before
FS. In my case, I keep registrations off of FS so it's only handling calls.
If you have registrations in FS, you can likely just have a Kamailio edge
proxy for advertising SIP outbound support, and have it proxy all traffic
into FS.

With this setup, FS will receive SIP traffic from Kamailio, and advertise
(in the SDP) its public IP address for RTP media (which needs to be allowed
through the firewall). Freeswitch will then open what it refers to as an
auto-adjust window for the RTP media. In other words, FS will assume that
the first address/port to send RTP media to the RTP port configured for a
call is the remote client for that call. As a result, FS is able to cope
with clients behind NAT on the media side as well. I believe this feature
is enabled by default, but you may have to enable it - you'd have to check
the docs on this one.

With those two pieces combined you should be able to get past any NAT
issues without the need for STUN/TURN. Unless you bypass media on FS, in
which case you're going to need those.

Hopefully that helps you out a bit.

Best,
Colin

On Thu, Jun 30, 2016 at 8:52 AM Rick Jarvis <rick.jarvis at magicmail.mooo.com>
wrote:

> I?d be interested to hear what different people use to provide some level
> of security for remote end-users such as homeworkers, and to get round NAT
> issues.
>
> We currently use OpenVPN, as this is built into the firmware of Yealink
> handsets (it?s a great feature, I?m not sure why more handset manufacturers
> don?t do this?!). The pros are that not only is it secure, but it also
> removes any problems with NAT for the RTP streams.
>
> The downsides are that it is complicated (and downright frustrating
> sometimes) to set up, and there are additional things to consider such as
> the server configuration and overheads.
>
> TLS/SSL with SRTP is another option, but my understanding of this is that
> it can cause NAT problems, with FreeSWITCH trying to initiate control
> channels back to the phone for inbound calls. In fact, I?ve always had
> problems with getting phones to work when behind NAT anyway, even without
> SSL/TLS. STUN can be used to ascertain the IP, but how do you handle
> situations where multiple handsets are behind NAT - you can?t open all RTP
> ports to all handsets at once?!!
>
> Would be very interested to hear thoughts and methods on these points.
>
> Thanks
> R
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From rick.jarvis at magicmail.mooo.com  Thu Jun 30 21:14:25 2016
From: rick.jarvis at magicmail.mooo.com (Rick Jarvis)
Date: Thu, 30 Jun 2016 18:14:25 +0100
Subject: [Freeswitch-users] Security vs compatibility / NAT etc
In-Reply-To: <CAPtU-UqLTYJjUj6iKQkUPt5TqFq5dWqxoq4NhVqLVi7ytOESKA@mail.gmail.com>
References: <C39782AE-E274-4417-A186-B042688E86EB@magicmail.mooo.com>
	<CAPtU-UqLTYJjUj6iKQkUPt5TqFq5dWqxoq4NhVqLVi7ytOESKA@mail.gmail.com>
Message-ID: <325EC6F6-7B59-4AF0-BCEE-F8BE30D3C523@magicmail.mooo.com>

That?s really helpful, thank you Colin! Yes, the prime concern is handling NAT for the remote phones, so am just starting to look into Kamailio now. A lot to get my head around, if only it was a FreeSWITCH module! ;)

So WRT to the security element, am I correct in thinking that people don?t generally worry about anyone piecing together unencrypted phone calls out on the net somewhere? Just worried that it might be risk that blows up in my face one day??

> On 30 Jun 2016, at 14:39, Colin Morelli <colin.morelli at gmail.com> wrote:
> 
> Rick,
> 
> (Sorry for the long email, hopefully it's helpful)
> 
> It sounds like you're mostly concerned with FS initiating calls to handsets behind NAT, is that correct?
> 
> If so, what you probably want is SIP outbound (RFC 5626). It's the best way to avoid NAT issues with clients. Under this model, clients keep a persistent connection open to the server. The server is responsible for using that connection to deliver INVITEs to the client, thus avoiding the need to ever open its own connection.
> 
> In my (relatively limited) experience with FS, it was able to act like a SIP outbound server, but it doesn't directly advertise it and supporting SIP outbound is really outside of the core scope of what FS does. So, in my setup, I use Kamailio to provide the SIP outbound support. A brief description of my setup (which seems to work fine with clients behind NAT)
> 
> Kamailio edge proxy cluster (provides SIP outbound support to clients, allows public SIP traffic)
> Kamailio proxy + registrar (only allows SIP traffic from inside the local network, provides registration support)
> Freeswitch (only allows SIP traffic from inside the local network, has a public IP address and open firewall for RTP traffic).
> 
> So, a registration from a client hits the Kamailio edge proxy, which parks the socket connection and sends it on to the second Kamailio proxy/registrar. When FS needs to make outbound calls to clients, it hits the Kamailio proxy/registrar, which forwards it to the edge proxy that has an existing connection the client and uses it to deliver the invite (this is all handled by Kamailio with it's outbound, path, registrar, and usrloc modules).
> 
> Note your setup might not require the use of two layers of proxies before FS. In my case, I keep registrations off of FS so it's only handling calls. If you have registrations in FS, you can likely just have a Kamailio edge proxy for advertising SIP outbound support, and have it proxy all traffic into FS.
> 
> With this setup, FS will receive SIP traffic from Kamailio, and advertise (in the SDP) its public IP address for RTP media (which needs to be allowed through the firewall). Freeswitch will then open what it refers to as an auto-adjust window for the RTP media. In other words, FS will assume that the first address/port to send RTP media to the RTP port configured for a call is the remote client for that call. As a result, FS is able to cope with clients behind NAT on the media side as well. I believe this feature is enabled by default, but you may have to enable it - you'd have to check the docs on this one.
> 
> With those two pieces combined you should be able to get past any NAT issues without the need for STUN/TURN. Unless you bypass media on FS, in which case you're going to need those.
> 
> Hopefully that helps you out a bit.
> 
> Best,
> Colin
> 
> On Thu, Jun 30, 2016 at 8:52 AM Rick Jarvis <rick.jarvis at magicmail.mooo.com <mailto:rick.jarvis at magicmail.mooo.com>> wrote:
> I?d be interested to hear what different people use to provide some level of security for remote end-users such as homeworkers, and to get round NAT issues.
> 
> We currently use OpenVPN, as this is built into the firmware of Yealink handsets (it?s a great feature, I?m not sure why more handset manufacturers don?t do this?!). The pros are that not only is it secure, but it also removes any problems with NAT for the RTP streams.
> 
> The downsides are that it is complicated (and downright frustrating sometimes) to set up, and there are additional things to consider such as the server configuration and overheads.
> 
> TLS/SSL with SRTP is another option, but my understanding of this is that it can cause NAT problems, with FreeSWITCH trying to initiate control channels back to the phone for inbound calls. In fact, I?ve always had problems with getting phones to work when behind NAT anyway, even without SSL/TLS. STUN can be used to ascertain the IP, but how do you handle situations where multiple handsets are behind NAT - you can?t open all RTP ports to all handsets at once?!!
> 
> Would be very interested to hear thoughts and methods on these points.
> 
> Thanks
> R
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>_________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From colin.morelli at gmail.com  Thu Jun 30 21:25:07 2016
From: colin.morelli at gmail.com (Colin Morelli)
Date: Thu, 30 Jun 2016 17:25:07 +0000
Subject: [Freeswitch-users] Security vs compatibility / NAT etc
In-Reply-To: <325EC6F6-7B59-4AF0-BCEE-F8BE30D3C523@magicmail.mooo.com>
References: <C39782AE-E274-4417-A186-B042688E86EB@magicmail.mooo.com>
	<CAPtU-UqLTYJjUj6iKQkUPt5TqFq5dWqxoq4NhVqLVi7ytOESKA@mail.gmail.com>
	<325EC6F6-7B59-4AF0-BCEE-F8BE30D3C523@magicmail.mooo.com>
Message-ID: <CAPtU-UrXAbbMDw=KCQRBE+ocqyeo06XZB1fnSszX5at7C8nf7Q@mail.gmail.com>

If you're going to route calls over the public internet I *highly*
recommend using TLS and SRTP. Generally speaking the computational overhead
for most encryption is negligible these days compared to the value and
safety it provides to you and your users.

For convenience it would be reasonable to terminate TLS in Kamailio and
send unencrypted SIP traffic to FS over the local network. RTP streams are
easy to secure as long as clients support it.

Are there concerns you have with enabling TLS+SRTP?

Best,
Colin
On Thu, Jun 30, 2016 at 1:18 PM Rick Jarvis <rick.jarvis at magicmail.mooo.com>
wrote:

> That?s really helpful, thank you Colin! Yes, the prime concern is handling
> NAT for the remote phones, so am just starting to look into Kamailio now. A
> lot to get my head around, if only it was a FreeSWITCH module! ;)
>
> So WRT to the security element, am I correct in thinking that people don?t
> generally worry about anyone piecing together unencrypted phone calls out
> on the net somewhere? Just worried that it might be risk that blows up in
> my face one day??
>
> On 30 Jun 2016, at 14:39, Colin Morelli <colin.morelli at gmail.com> wrote:
>
> Rick,
>
> (Sorry for the long email, hopefully it's helpful)
>
> It sounds like you're mostly concerned with FS initiating calls to
> handsets behind NAT, is that correct?
>
> If so, what you probably want is SIP outbound (RFC 5626). It's the best
> way to avoid NAT issues with clients. Under this model, clients keep a
> persistent connection open to the server. The server is responsible for
> using that connection to deliver INVITEs to the client, thus avoiding the
> need to ever open its own connection.
>
> In my (relatively limited) experience with FS, it was able to act like a
> SIP outbound server, but it doesn't directly advertise it and supporting
> SIP outbound is really outside of the core scope of what FS does. So, in my
> setup, I use Kamailio to provide the SIP outbound support. A brief
> description of my setup (which seems to work fine with clients behind NAT)
>
> Kamailio edge proxy cluster (provides SIP outbound support to clients,
> allows public SIP traffic)
> Kamailio proxy + registrar (only allows SIP traffic from inside the local
> network, provides registration support)
> Freeswitch (only allows SIP traffic from inside the local network, has a
> public IP address and open firewall for RTP traffic).
>
> So, a registration from a client hits the Kamailio edge proxy, which parks
> the socket connection and sends it on to the second Kamailio
> proxy/registrar. When FS needs to make outbound calls to clients, it hits
> the Kamailio proxy/registrar, which forwards it to the edge proxy that has
> an existing connection the client and uses it to deliver the invite (this
> is all handled by Kamailio with it's outbound, path, registrar, and usrloc
> modules).
>
> Note your setup might not require the use of two layers of proxies before
> FS. In my case, I keep registrations off of FS so it's only handling calls.
> If you have registrations in FS, you can likely just have a Kamailio edge
> proxy for advertising SIP outbound support, and have it proxy all traffic
> into FS.
>
> With this setup, FS will receive SIP traffic from Kamailio, and advertise
> (in the SDP) its public IP address for RTP media (which needs to be allowed
> through the firewall). Freeswitch will then open what it refers to as an
> auto-adjust window for the RTP media. In other words, FS will assume that
> the first address/port to send RTP media to the RTP port configured for a
> call is the remote client for that call. As a result, FS is able to cope
> with clients behind NAT on the media side as well. I believe this feature
> is enabled by default, but you may have to enable it - you'd have to check
> the docs on this one.
>
> With those two pieces combined you should be able to get past any NAT
> issues without the need for STUN/TURN. Unless you bypass media on FS, in
> which case you're going to need those.
>
> Hopefully that helps you out a bit.
>
> Best,
> Colin
>
> On Thu, Jun 30, 2016 at 8:52 AM Rick Jarvis <
> rick.jarvis at magicmail.mooo.com> wrote:
>
>> I?d be interested to hear what different people use to provide some level
>> of security for remote end-users such as homeworkers, and to get round NAT
>> issues.
>>
>> We currently use OpenVPN, as this is built into the firmware of Yealink
>> handsets (it?s a great feature, I?m not sure why more handset manufacturers
>> don?t do this?!). The pros are that not only is it secure, but it also
>> removes any problems with NAT for the RTP streams.
>>
>> The downsides are that it is complicated (and downright frustrating
>> sometimes) to set up, and there are additional things to consider such as
>> the server configuration and overheads.
>>
>> TLS/SSL with SRTP is another option, but my understanding of this is that
>> it can cause NAT problems, with FreeSWITCH trying to initiate control
>> channels back to the phone for inbound calls. In fact, I?ve always had
>> problems with getting phones to work when behind NAT anyway, even without
>> SSL/TLS. STUN can be used to ascertain the IP, but how do you handle
>> situations where multiple handsets are behind NAT - you can?t open all RTP
>> ports to all handsets at once?!!
>>
>> Would be very interested to hear thoughts and methods on these points.
>>
>> Thanks
>> R
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From mike at jerris.com  Thu Jun 30 21:32:26 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 30 Jun 2016 13:32:26 -0400
Subject: [Freeswitch-users] Security vs compatibility / NAT etc
In-Reply-To: <325EC6F6-7B59-4AF0-BCEE-F8BE30D3C523@magicmail.mooo.com>
References: <C39782AE-E274-4417-A186-B042688E86EB@magicmail.mooo.com>
	<CAPtU-UqLTYJjUj6iKQkUPt5TqFq5dWqxoq4NhVqLVi7ytOESKA@mail.gmail.com>
	<325EC6F6-7B59-4AF0-BCEE-F8BE30D3C523@magicmail.mooo.com>
Message-ID: <91CCB5AF-3D64-416E-854F-3FEB2E4FB4A4@jerris.com>

People generally don't worry about it... incorrectly.. its quite trivial to go from a pcap to a full audio recording of a call, so anyone in the call path can listen.  This is why newer protocols such as webrtc have security features required to use it at all, as defaulting secure is MUCH better.

> On Jun 30, 2016, at 1:14 PM, Rick Jarvis <rick.jarvis at magicmail.mooo.com> wrote:
> 
> That?s really helpful, thank you Colin! Yes, the prime concern is handling NAT for the remote phones, so am just starting to look into Kamailio now. A lot to get my head around, if only it was a FreeSWITCH module! ;)
> 
> So WRT to the security element, am I correct in thinking that people don?t generally worry about anyone piecing together unencrypted phone calls out on the net somewhere? Just worried that it might be risk that blows up in my face one day??
> 
>> On 30 Jun 2016, at 14:39, Colin Morelli <colin.morelli at gmail.com <mailto:colin.morelli at gmail.com>> wrote:
>> 
>> Rick,
>> 
>> (Sorry for the long email, hopefully it's helpful)
>> 
>> It sounds like you're mostly concerned with FS initiating calls to handsets behind NAT, is that correct?
>> 
>> If so, what you probably want is SIP outbound (RFC 5626). It's the best way to avoid NAT issues with clients. Under this model, clients keep a persistent connection open to the server. The server is responsible for using that connection to deliver INVITEs to the client, thus avoiding the need to ever open its own connection.
>> 
>> In my (relatively limited) experience with FS, it was able to act like a SIP outbound server, but it doesn't directly advertise it and supporting SIP outbound is really outside of the core scope of what FS does. So, in my setup, I use Kamailio to provide the SIP outbound support. A brief description of my setup (which seems to work fine with clients behind NAT)
>> 
>> Kamailio edge proxy cluster (provides SIP outbound support to clients, allows public SIP traffic)
>> Kamailio proxy + registrar (only allows SIP traffic from inside the local network, provides registration support)
>> Freeswitch (only allows SIP traffic from inside the local network, has a public IP address and open firewall for RTP traffic).
>> 
>> So, a registration from a client hits the Kamailio edge proxy, which parks the socket connection and sends it on to the second Kamailio proxy/registrar. When FS needs to make outbound calls to clients, it hits the Kamailio proxy/registrar, which forwards it to the edge proxy that has an existing connection the client and uses it to deliver the invite (this is all handled by Kamailio with it's outbound, path, registrar, and usrloc modules).
>> 
>> Note your setup might not require the use of two layers of proxies before FS. In my case, I keep registrations off of FS so it's only handling calls. If you have registrations in FS, you can likely just have a Kamailio edge proxy for advertising SIP outbound support, and have it proxy all traffic into FS.
>> 
>> With this setup, FS will receive SIP traffic from Kamailio, and advertise (in the SDP) its public IP address for RTP media (which needs to be allowed through the firewall). Freeswitch will then open what it refers to as an auto-adjust window for the RTP media. In other words, FS will assume that the first address/port to send RTP media to the RTP port configured for a call is the remote client for that call. As a result, FS is able to cope with clients behind NAT on the media side as well. I believe this feature is enabled by default, but you may have to enable it - you'd have to check the docs on this one.
>> 
>> With those two pieces combined you should be able to get past any NAT issues without the need for STUN/TURN. Unless you bypass media on FS, in which case you're going to need those.
>> 
>> Hopefully that helps you out a bit.
>> 
>> Best,
>> Colin
>> 
>> On Thu, Jun 30, 2016 at 8:52 AM Rick Jarvis <rick.jarvis at magicmail.mooo.com <mailto:rick.jarvis at magicmail.mooo.com>> wrote:
>> I?d be interested to hear what different people use to provide some level of security for remote end-users such as homeworkers, and to get round NAT issues.
>> 
>> We currently use OpenVPN, as this is built into the firmware of Yealink handsets (it?s a great feature, I?m not sure why more handset manufacturers don?t do this?!). The pros are that not only is it secure, but it also removes any problems with NAT for the RTP streams.
>> 
>> The downsides are that it is complicated (and downright frustrating sometimes) to set up, and there are additional things to consider such as the server configuration and overheads.
>> 
>> TLS/SSL with SRTP is another option, but my understanding of this is that it can cause NAT problems, with FreeSWITCH trying to initiate control channels back to the phone for inbound calls. In fact, I?ve always had problems with getting phones to work when behind NAT anyway, even without SSL/TLS. STUN can be used to ascertain the IP, but how do you handle situations where multiple handsets are behind NAT - you can?t open all RTP ports to all handsets at once?!!
>> 
>> Would be very interested to hear thoughts and methods on these points.
>> 
>> Thanks
>> R
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
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>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
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>> 
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>> http://www.freeswitch.org <http://www.freeswitch.org/>_________________________________________________________________________
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>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
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>> 
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> 
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
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> 
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From danny.gershman at gmail.com  Thu Jun 30 21:49:05 2016
From: danny.gershman at gmail.com (Danny Gershman)
Date: Thu, 30 Jun 2016 17:49:05 +0000
Subject: [Freeswitch-users] Docker DTMF related issue
Message-ID: <CAEorZynB+FfXjxTe=BCZHxPpTv=aod9cwXSk3bYuiN5q5z6jbw@mail.gmail.com>

I'm having an issue with my DTMF to a Docker container running FS 1.6.9.
My sofia profile is set up to use RFC2833.  However from a softphone and
another FS server it's not detecting the digits.  If I switch to INFO
method, it works fine.

Also is there a quick was to do a uuid_send_dtmf using SIP INFO to work
around this issue temporarily?  I saw something about a channel variable
dtmf_type, which didn't seem to work when I did a uuid_setvar.

Thanks,
Danny Gershman
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