From bote_radio at botecomm.com  Mon Feb  1 00:11:12 2016
From: bote_radio at botecomm.com (Bote Man)
Date: Sun, 31 Jan 2016 16:11:12 -0500
Subject: [Freeswitch-users] [Video Calling Issue][Freeswitch 1.7]
In-Reply-To: <CAJUJwtj9U4gjMa=H84YemyfyPaeO=Nd9p2XwbA=T+nSE1UE2AQ@mail.gmail.com>
References: <CADSVerCYq51oAosRtGbThyqanwymaUOL1AuUW9wG92u2uB6K6A@mail.gmail.com>
	<CAJUJwtj9U4gjMa=H84YemyfyPaeO=Nd9p2XwbA=T+nSE1UE2AQ@mail.gmail.com>
Message-ID: <00f801d15c6b$ea5bdd00$bf139700$@botecomm.com>

Pastebin is preferred to posting miles of log files.

 

https://pastebin.freeswitch.org/

 

Select ?FreeSWITCH log? format for easier troubleshooting.

 

Thanks.

 

 

---

Bote

 

FreeSWITCH Docs Janitor

 <http://freeswitch.org/confluence> http://freeswitch.org/confluence

 

 

 

 

From: SamyGo
Sent: Sunday, 31 January, 2016 14:48
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [Video Calling Issue][Freeswitch 1.7]

 

Hi Bilal,
I have FS 1.7 and I get video calls going on it without trouble. Could you post the CLI logs as well as the SIP trace appearing for the  all..only the First INVITE from caller and 200OK should suffice. 
Can you tell if you either mod_h26x or mod_av loaded in FS. Also load mod_vpx and then try making calls. 
FS 1.6 is now part of newer versions so dont need to explicitly install 1m6 for video only. 

Regards,
Sammy 

On Jan 30, 2016 12:09, "Bilal Abbasi" <bilaln018 at gmail.com> wrote:

Hi Users,

 

I am writing this question after reading old questions and forums, so my problem is i am unable to dial video call using freeswitch 1.7.

I have enabled video codecs in var.xml (VP8,H263,H264) and i am using bria sipfone for UA1 and UA2.

I am able to dial the normal call but unable to dial video.

do i need any further variables to activate video calling?

One of my friend told me to shift to FS 1.6 as that provides transcoding.

so can anybody help me out.

 

FS Version: 1.7.0

OS: jessie 8

Softphone: Bria

 

Regards

Abbasi 


_________________________________________________________________________



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From omortimer at gmail.com  Mon Feb  1 00:58:22 2016
From: omortimer at gmail.com (Oz Mortimer)
Date: Sun, 31 Jan 2016 21:58:22 +0000
Subject: [Freeswitch-users] extra header account code is not written to
	cdr if cancel is received a few ms after invite
In-Reply-To: <3CBD15A5-819B-487B-9DE0-C8120DDCAC94@gmail.com>
References: <56AE42E1.3080703@gmail.com>
	<3CBD15A5-819B-487B-9DE0-C8120DDCAC94@gmail.com>
Message-ID: <3D075DBF-8FEF-40CE-9922-A7CFC462512E@gmail.com>

Try export rather than set

> On 31 Jan 2016, at 18:45, servtelar at gmail.com wrote:
> 
> Shouldn't that be done as inline?
> 
> Sent from my iPhone
> 
>> On Jan 31, 2016, at 12:22 PM, Panagiotis Skoulikaritis <pskoul at gmail.com> wrote:
>> 
>> Dear all
>> 
>> I have an implementation FreeSWITCH as a sort of SBC, it is used to send
>> the calls to the terminating carriers and do topology hiding, nothing
>> fancy. Also I gather cdrs from the FreeSWITCH.
>> 
>> In order to distinguish each customer on the FS cdrs I send an extra
>> header containing the accountcode.
>> 
>> I have noticed that if the call is canceled immediately on the same sec,
>> the account code is not written on the cdr.
>> To be more precise the cancel is send a few milliseconds after it has
>> received the invite, and before the FreeSWITCH has sent the call to the
>> terminating carrier (I'm using Homer Sipcapture to capture all the
>> traces and I don't see an attempt being made at the terminating carrier)
>> also I don't see a b-leg cdr.
>> 
>> FreeSWITCH is writing both a-leg and b-leg cdrs in csv format.
>> 
>> The dialplan that I use is simple
>> 
>> <extension name="carrier-1">
>> <condition field="network_addr"
>> expression="^(^xx\.xx\.xx\.xx|^yy\.yy\.yy\.yy)$"/>
>> <condition field="destination_number" expression="^2#(\d+)$">
>> <action application="set" data="call_direction=outgoing"/>
>> <action application="set" data="hangup_after_bridge=true"/>
>> <action application="set" data="continue_on_fail=true"/>
>> <action application="set" data="sip_cid_in_1xx=false"/>
>> <action application="set" data="inherit_codec=true"/>
>> <action application="set" data="accountcode=${sip_h_X-ACCOUNTCODE}"/>
>> <action application="unset" data="sip_h_X-ACCOUNTCODE"/>
>> <action application="set" data="execute_on_answer=sched_hangup +21600
>> alloted_timeout"/>
>> <action application="bridge" data="sofia/external/4567$1 at zz.zz.zz.zz:5070"/>
>> </condition>
>> </extension>
>> 
>> 
>> any idea how I can make sure that the account code will always be written ?
>> 
>> 
>> Best Regards
>> 
>> Panagiotis
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: 
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From bilaln018 at gmail.com  Mon Feb  1 08:08:46 2016
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Mon, 1 Feb 2016 10:08:46 +0500
Subject: [Freeswitch-users] [Video Calling Issue][Freeswitch 1.7]
In-Reply-To: <CAJUJwtj9U4gjMa=H84YemyfyPaeO=Nd9p2XwbA=T+nSE1UE2AQ@mail.gmail.com>
References: <CADSVerCYq51oAosRtGbThyqanwymaUOL1AuUW9wG92u2uB6K6A@mail.gmail.com>
	<CAJUJwtj9U4gjMa=H84YemyfyPaeO=Nd9p2XwbA=T+nSE1UE2AQ@mail.gmail.com>
Message-ID: <CADSVerCm=s=G27aDv3iVvYhbrdTDDYhRMWH6ZSwv8p6FFBqfZg@mail.gmail.com>

Hi sammy,
Thanks for your reply, yes i have enabled mod_h26x and mod_vpx loaded in
modules.conf as well.i have enabled mod_av as well.
ok i will share the logs and dump file.

Regards
Abbasi


On Mon, Feb 1, 2016 at 12:47 AM, SamyGo <govoiper at gmail.com> wrote:

> Hi Bilal,
> I have FS 1.7 and I get video calls going on it without trouble. Could you
> post the CLI logs as well as the SIP trace appearing for the  all..only the
> First INVITE from caller and 200OK should suffice.
> Can you tell if you either mod_h26x or mod_av loaded in FS. Also load
> mod_vpx and then try making calls.
> FS 1.6 is now part of newer versions so dont need to explicitly install
> 1m6 for video only.
>
> Regards,
> Sammy
> On Jan 30, 2016 12:09, "Bilal Abbasi" <bilaln018 at gmail.com> wrote:
>
>> Hi Users,
>>
>> I am writing this question after reading old questions and forums, so my
>> problem is i am unable to dial video call using freeswitch 1.7.
>> I have enabled video codecs in var.xml (VP8,H263,H264) and i am using
>> bria sipfone for UA1 and UA2.
>> I am able to dial the normal call but unable to dial video.
>> do i need any further variables to activate video calling?
>> One of my friend told me to shift to FS 1.6 as that provides transcoding.
>> so can anybody help me out.
>>
>> FS Version: 1.7.0
>> OS: jessie 8
>> Softphone: Bria
>>
>> Regards
>> Abbasi
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From luis.azedo at factorlusitano.com  Mon Feb  1 16:47:22 2016
From: luis.azedo at factorlusitano.com (Luis Azedo)
Date: Mon, 1 Feb 2016 13:47:22 +0000
Subject: [Freeswitch-users] recording problems with mod_shout debian/master
Message-ID: <CAEvFzR8Xxu57=rLnbwPkMJFAGMJXn7O2cik41YuFv128BcwZ9g@mail.gmail.com>

Hi,

anyone having problems recording in mp3  format ?

2016-02-01 13:41:47.108078 [INFO] kazoo_node.c:625 exec:
*uuid_record(96722133-5060-508 at BJC.BGI.CG.BD
<96722133-5060-508 at BJC.BGI.CG.BD> start
/tmp/878202b3a78863ec40f2dbc800fcde66.mp3 600)*
2016-02-01 13:41:47.108078 [DEBUG]* switch_core_file.c:323 File
/tmp/878202b3a78863ec40f2dbc800fcde66.mp3 sample rate 8000 doesn't match
requested rate 16000*
2016-02-01 13:41:47.128429 [DEBUG] *switch_core_media_bug.c:828 Attaching
BUG to sofia/sipinterface_1/995582142 at teste.sip.90e9.com
<995582142 at teste.sip.90e9.com>*
2016-02-01 13:41:47.148020 [DEBUG] *switch_ivr_async.c:1491 No silence
detection configured; assuming start of speech*
2016-02-01 13:41:47.168888 [INFO] *mod_shout.c:326 LAME 3.99.5 64bits
(http://lame.sf.net <http://lame.sf.net>)*
2016-02-01 13:41:47.168888 [INFO] *mod_shout.c:326 polyphase lowpass filter
disabled*
2016-02-01 13:41:47.168888 [ERR] *mod_shout.c:1057 MP3 encode error -1!*
2016-02-01 13:41:47.168888 [ERR] *switch_ivr_async.c:1160 Error writing
/tmp/878202b3a78863ec40f2dbc800fcde66.mp3*
2016-02-01 13:41:47.168888 [ERR] *mod_shout.c:1057 MP3 encode error -1!*
2016-02-01 13:41:47.168888 [ERR] *switch_ivr_async.c:1160 Error writing
/tmp/878202b3a78863ec40f2dbc800fcde66.mp3*
2016-02-01 13:41:47.688430 [ERR] *mod_shout.c:1057 MP3 encode error -1!*
2016-02-01 13:41:47.688430 [ERR] *switch_ivr_async.c:1160 Error writing
/tmp/878202b3a78863ec40f2dbc800fcde66.mp3*
2016-02-01 13:41:47.688430 [ERR] *mod_shout.c:1057 MP3 encode error -1!*
2016-02-01 13:41:47.688430 [ERR] *switch_ivr_async.c:1160 Error writing
/tmp/878202b3a78863ec40f2dbc800fcde66.mp3*
2016-02-01 13:41:47.688430 [ERR] *mod_shout.c:1057 MP3 encode error -1!*
2016-02-01 13:41:47.688430 [ERR] *switch_ivr_async.c:1160 Error writing
/tmp/878202b3a78863ec40f2dbc800fcde66.mp3*

Thanks
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From tg at level5.de  Mon Feb  1 17:12:25 2016
From: tg at level5.de (=?UTF-8?Q?Thorsten_G=c3=b6llner?=)
Date: Mon, 1 Feb 2016 15:12:25 +0100
Subject: [Freeswitch-users] Verto vs. SIP.js
Message-ID: <56AF67C9.2010700@level5.de>

Hi,

I want to setup a Click-2-Call-Button for our website. Is there any 
significant difference between Verto-Mod and libraries such as SIP.js?

I do not really understand the advantage of Verto (if there is any).

Thanks in advance,

Thorsten


From yadenis at seznam.cz  Mon Feb  1 17:25:39 2016
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Mon, 1 Feb 2016 15:25:39 +0100
Subject: [Freeswitch-users] Verto vs. SIP.js
In-Reply-To: <56AF67C9.2010700@level5.de>
References: <56AF67C9.2010700@level5.de>
Message-ID: <95599933.20160201152539@seznam.cz>

Dobr? den,

I use sip.js. It turned out to be a lot easier that Verto.

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

pond?l? 1. ?nora 2016, 15:12:25, napsal jste:

> Hi,

> I want to setup a Click-2-Call-Button for our website. Is there any 
> significant difference between Verto-Mod and libraries such as SIP.js?

> I do not really understand the advantage of Verto (if there is any).

> Thanks in advance,

> Thorsten

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

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

> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From DEdwards at vertical.com  Mon Feb  1 17:34:42 2016
From: DEdwards at vertical.com (Dan Edwards)
Date: Mon, 1 Feb 2016 14:34:42 +0000
Subject: [Freeswitch-users] WebSocket behind NGINX
In-Reply-To: <56AD0CE7.6000607@gmail.com>
References: <56AD0CE7.6000607@gmail.com>
Message-ID: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C997A4E4@PHXEX2.vertical.com>

I'm also running behind Nginx and what I found worked was to proxy to the actual IP address (192.168.1.1 vs. 127.0.0.1), then explicitly removing 192.168.1.1 from the localnet ACL in acl.conf. I had to remove 192.168.1.1 from localnet so FS will offer external IP addresses for RTP. 


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anton
Sent: Saturday, January 30, 2016 2:20 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] WebSocket behind NGINX

Hello All,

I have to proxy all websocket requests though a nginx server. Right now I am using next configuration:

map $http_upgrade $connection_upgrade {
     default upgrade;
     ''      close;
}

server {
     listen 443;
     server_name  wss.somedomain.com.ua;

     ssl on;
     ssl_certificate      /etc/nginx/cert.pem;
     ssl_certificate_key  /etc/nginx/private.key;

     location / {
         proxy_pass http://127.0.0.1:5066;
         proxy_http_version 1.1;
         proxy_set_header Upgrade $http_upgrade;
         proxy_set_header Connection $connection_upgrade;
         proxy_read_timeout 86400s;
    }

     access_log /var/log/nginx/wss_access;
     error_log /var/log/nginx/wss_error debug; }

I dumped traffic from nginx and found out that "switching protocol" 
phrase was successful but INVITE message from my browser in pending state.
Maybe FreeSWITCH wants real IP not loopback? Who have faced with similar problem?

BR,
Anton

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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at 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 bote_radio at botecomm.com  Mon Feb  1 17:42:48 2016
From: bote_radio at botecomm.com (Bote Man)
Date: Mon, 1 Feb 2016 09:42:48 -0500
Subject: [Freeswitch-users] Verto vs. SIP.js
In-Reply-To: <95599933.20160201152539@seznam.cz>
References: <56AF67C9.2010700@level5.de> <95599933.20160201152539@seznam.cz>
Message-ID: <003a01d15cfe$d26c3350$774499f0$@botecomm.com>

Denis, what difficulties did you experience with Verto?

 

Thanks.

 

Bote

 

 

From: Denis Jakovlev
Sent: Monday, 01 February, 2016 09:26
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Verto vs. SIP.js

 

Dobr? den,

I use sip.js. It turned out to be a lot easier that Verto.

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

pond?l? 1. ?nora 2016, 15:12:25, napsal jste:

> Hi,

> I want to setup a Click-2-Call-Button for our website. Is there any 
> significant difference between Verto-Mod and libraries such as SIP.js?

> I do not really understand the advantage of Verto (if there is any).

> Thanks in advance,

> Thorsten

> _________________________________________________________________________



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From eschmidbauer at gmail.com  Mon Feb  1 18:16:45 2016
From: eschmidbauer at gmail.com (E. Schmidbauer)
Date: Mon, 1 Feb 2016 10:16:45 -0500
Subject: [Freeswitch-users] FreeSWITCH in virtual environments
In-Reply-To: <58E2C5C4-87FF-4E01-8048-6C39B9362B02@gmail.com>
References: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C9979B74@PHXEX2.vertical.com>
	<CAJ0sghSaQhKgD4X57DkEGP_MBxHgHi1vTLDh7E7tQXZtKRPz_A@mail.gmail.com>
	<88218521-FB01-4698-9DBF-625D99E8BC79@gmail.com>
	<CAHtxdDfjiAeSEZfnFSoAAK3syGMwBfFrV=Ox+np8AfdTteuqeg@mail.gmail.com>
	<58E2C5C4-87FF-4E01-8048-6C39B9362B02@gmail.com>
Message-ID: <CAJwe+CBc64_0efV=-0RNr3kB_gtPPU6vTdbcxKWL7m+wGsmbVw@mail.gmail.com>

Can anyone share the VM memory/cpu specs used in these cases?
We want to run around 300 CPS on FS (running on vmware).
very little transcoding (if any), audio only
How much memory/cpu should be provisioned?
I see Grant mentioned "single VM host (6 Cores, 12 Threads)" but how much
memory?
Thanks,
E

On Fri, Jan 29, 2016 at 10:00 PM, servtelar <servtelar at gmail.com> wrote:

> Thanks a lot guys for sharing this info. It?s really helpful.
>
>
> On Jan 28, 2016, at 6:18 PM, Sergey Safarov <s.safarov at gmail.com> wrote:
>
> We have to core ESXi vm with 140 session (70 calls) with have 70 CPU load.
>
> Sergey
>
> On Fri, Jan 29, 2016 at 2:01 AM, servtelar <servtelar at gmail.com> wrote:
>
>> Hi Chad
>>
>> How many legs you are handling with 20 cores on a conference?
>>
>> Regards
>>
>> Gustavo
>>
>> On Jan 28, 2016, at 7:55 PM, Chad Phillips <chad at apartmentlines.com>
>> wrote:
>>
>> I've had very good luck running the newer video branch code on
>> ProfitBricks: https://www.profitbricks.com/
>>
>> As far as I understand, the CPU cycles are guaranteed on their platform.
>> I've had to put as many as 20 cores on a server to handle some of our
>> busier video conference calls, but with that it runs quite smoothly.
>>
>> On Thu, Jan 28, 2016 at 2:15 PM, Dan Edwards <DEdwards at vertical.com>
>> wrote:
>>
>>> I am reviewing the Confluence Virtualization page and had some
>>> questions, in particular about VMWare. My company distributes some of its
>>> software as a VMWare image file and we were looking to distribute a new
>>> product using FS in the same manner. The products operate at a customer
>>> premise, on their VMWare infrastructure, not in a cloud environment. Since
>>> our customers already have VMWare, switching to a different VM
>>> infrastructure is going to hurt, so I am looking for options/alternatives.
>>>
>>> First, does anybody know if the virtual timing issues with VMWare have
>>> improved since this page was last updated in 2014? Is VMWare still not good
>>> enough? Is it possible to throw CPU & memory at this and make VMWare good
>>> enough, or is the virtual timing just not workable?
>>>
>>> On the virtualization page, there was a comment from 2010 that you might
>>> be happy with a High CPU Medium instance on AWS EC2. Certainly workload is
>>> a factor here, but I am trying to get my head around how big a machine to
>>> perform how small a workload. Is there a place where people talk about
>>> their experiences?
>>>
>>> Are there other VM platforms that might be acceptable?
>>>
>>> Any help or comment is appreciated.
>>>
>>> Thank you,
>>> Dan
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 llribeiro90 at gmail.com  Mon Feb  1 18:38:25 2016
From: llribeiro90 at gmail.com (Leonardo Lima Ribeiro)
Date: Mon, 1 Feb 2016 13:38:25 -0200
Subject: [Freeswitch-users] Simple LuaScript to Record Session Failing
In-Reply-To: <CAJ0sghQriAOy6QORSpMQPn=-D6p6JxPDWPkSP3tzHfcBcrkdDg@mail.gmail.com>
References: <F47B06F3-9A8D-44ED-8936-7C4C0C9EDC52@gmail.com>
	<CAOiUWxEL3=u3vpqK1TPc6Ws91vkTZ02+BCx0gmyq+pX7e8Kk1Q@mail.gmail.com>
	<CDFC4F0A-FBDF-41AF-8FC4-7B9A188317D1@gmail.com>
	<CAJ0sghQriAOy6QORSpMQPn=-D6p6JxPDWPkSP3tzHfcBcrkdDg@mail.gmail.com>
Message-ID: <FE6AF8FC-E6D4-4011-84E4-DFA526188206@gmail.com>

Thanks Chad and Oz.

I tried two things:

1) Oz suggestion:

local new_session = freeswitch.Session("[origination_caller_id_name='7136694967',origination_caller_id_number='7136694967']sofia/gateway/4_NEXTIVA/3157244022", session);
new_session:setVariable(?RECORD_STEREO?,?false?)
new_session:sleep("5000")
new_session:streamFile("voicemail/vm-goodbye.wav")
new_session:sleep("10000")
new_session:hangup()

Did not work?

2) Chad suggestion:

local new_session = freeswitch.Session("[origination_caller_id_name='7136694967',origination_caller_id_number='7136694967']sofia/gateway/4_NEXTIVA/3157244022", session);

uuid = new_session:get_uuid()
api = freeswitch.API()
reply = api:executeString("uuid_record " .. uuid .. " start /usr/local/freeswitch/recordings/myrecording.wav")

new_session:sleep("5000")
new_session:streamFile("voicemail/vm-goodbye.wav")
new_session:sleep("10000")
new_session:hangup()

Did not work too? 

Both cases I have 5 seconds of silence (first 5000 pause) and then when I play the Good Bye wav file it starts to record normally. So I have 5 seconds of silence, good bye sound and 10 seconds of recording.

Chad, I can?t use session:recordFile because it?s a synchronous command that blocks my script.. I need to do a lot of actions like pauses, stream files etc., while the call is still going on?

Maybe should I open a ticket for FreeSWITCH support? (I even don?t know if it exists hehe, maybe a github issue?)

Thank you,


> On Jan 30, 2016, at 12:36 AM, Chad Phillips <chad at apartmentlines.com> wrote:
> 
> have you tried the session method specifically for recording files?
> 
> https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:recordFile <https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:recordFile>
> 
> or if that doesn't work, maybe this:
> 
> local new_session = freeswitch.Session("someoriginatestring", session);
> uuid = new_session:get_uuid()
> api = freeswitch.API()
> reply = api:executeString("uuid_record " .. uuid .. " start " .. filepath)
> 
> On Fri, Jan 29, 2016 at 1:31 PM, Oz Mortimer <omortimer at gmail.com <mailto:omortimer at gmail.com>> wrote:
> since this is a single left call try setting record_stereo to false - https://wiki.freeswitch.org/wiki/Variable_RECORD_STEREO <https://wiki.freeswitch.org/wiki/Variable_RECORD_STEREO>
> I'm probably wrong, but worth a shot!
> 
> On 29 Jan 2016, at 19:53, Leonardo Ribeiro <llribeiro90 at gmail.com <mailto:llribeiro90 at gmail.com>> wrote:
> 
>> Hello Guys,
>> 
>> Any idea?
>> I could not evolve this yet...
>> 
>> Thank you,
>> 
>> 2016-01-28 18:52 GMT-02:00 Leonardo Lima Ribeiro <llribeiro90 at gmail.com <mailto:llribeiro90 at gmail.com>>:
>> Hello all,
>> 
>> I?m trying to record an IVR using my gateway to do the outbound call in my luascript:
>> 
>> local new_session = freeswitch.Session("[origination_caller_id_name=?987654321',origination_caller_id_number=?987654321']sofia/gateway/MY_GATEWAY/3157244022 <tel:3157244022>", session);
>> new_session:execute("record_session","/usr/local/freeswitch/recordings/myrecording.wav")
>> new_session:sleep("10000")
>> new_session:hangup()
>> 
>> So in the above script I just call to the Bank Of America as an example and try to record the first 10 seconds of the call in the recordings path.
>> 
>> The problem is that I have an empty recording file.. Why?
>> 
>> The funny thing is: if I add this command after the record_session command: 
>> new_session:streamFile("voicemail/vm-goodbye.wav?);
>> 
>> And then this is the entire new script:
>> local new_session = freeswitch.Session("[origination_caller_id_name=?987654321',origination_caller_id_number=?987654321']sofia/gateway/MY_GATEWAY/3157244022 <tel:3157244022>", session);
>> new_session:execute("record_session","/usr/local/freeswitch/recordings/myrecording.wav?)
>> new_session:streamFile("voicemail/vm-goodbye.wav?);
>> new_session:sleep("10000")
>> new_session:hangup()
>> 
>> I can hear the Good Bye sound from my script and then hear the Bank of America IVR.
>> 
>> I just don?t understand why the record works if I play a sound in our side and the record does not work if I don?t play any sound.
>> 
>> Do you know what?s happening? How can I solve this?
>> 
>> Thank you!
>> 
>> 
>> 
>> 
>> -- 
>> Leonardo Lima Ribeiro
>> _________________________________________________________________________
>> 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 yadenis at seznam.cz  Mon Feb  1 18:30:13 2016
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Mon, 1 Feb 2016 16:30:13 +0100
Subject: [Freeswitch-users] Verto vs. SIP.js
In-Reply-To: <003a01d15cfe$d26c3350$774499f0$@botecomm.com>
References: <56AF67C9.2010700@level5.de> <95599933.20160201152539@seznam.cz> 
	<003a01d15cfe$d26c3350$774499f0$@botecomm.com>
Message-ID: <1105687706.20160201163013@seznam.cz>

Dobr? den,

i think this Verto its too much VendorLock :) 

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

pond?l? 1. ?nora 2016, 15:42:48, napsal jste:


Denis, what difficulties did you experience with Verto?
 
Thanks.
 
Bote
 
 
From: Denis Jakovlev
Sent: Monday, 01 February, 2016 09:26
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Verto vs. SIP.js
 
Dobr? den,

I use sip.js. It turned out to be a lot easier that Verto.

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

pond?l? 1. ?nora 2016, 15:12:25, napsal jste:

> Hi,

> I want to setup a Click-2-Call-Button for our website. Is there any 
> significant difference between Verto-Mod and libraries such as SIP.js?

> I do not really understand the advantage of Verto (if there is any).

> Thanks in advance,

> Thorsten

> _________________________________________________________________________
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From lexxua at gmail.com  Mon Feb  1 18:34:06 2016
From: lexxua at gmail.com (Volodymyr Fedorov)
Date: Mon, 1 Feb 2016 16:34:06 +0100
Subject: [Freeswitch-users] FreeSWITCH in virtual environments
In-Reply-To: <CAJwe+CBc64_0efV=-0RNr3kB_gtPPU6vTdbcxKWL7m+wGsmbVw@mail.gmail.com>
References: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C9979B74@PHXEX2.vertical.com>
	<CAJ0sghSaQhKgD4X57DkEGP_MBxHgHi1vTLDh7E7tQXZtKRPz_A@mail.gmail.com>
	<88218521-FB01-4698-9DBF-625D99E8BC79@gmail.com>
	<CAHtxdDfjiAeSEZfnFSoAAK3syGMwBfFrV=Ox+np8AfdTteuqeg@mail.gmail.com>
	<58E2C5C4-87FF-4E01-8048-6C39B9362B02@gmail.com>
	<CAJwe+CBc64_0efV=-0RNr3kB_gtPPU6vTdbcxKWL7m+wGsmbVw@mail.gmail.com>
Message-ID: <CAO_GGAoaFOg8cJ930DHbjyXKxZC1W6Oj7G5KajvjHm=4Gr5FdQ@mail.gmail.com>

Hello, it depends.
In my case 24 threads and 8gb of ram was quite enough. But I used only
xml-dialplan and hash counters.

On Mon, Feb 1, 2016 at 4:16 PM, E. Schmidbauer <eschmidbauer at gmail.com>
wrote:

> Can anyone share the VM memory/cpu specs used in these cases?
> We want to run around 300 CPS on FS (running on vmware).
> very little transcoding (if any), audio only
> How much memory/cpu should be provisioned?
> I see Grant mentioned "single VM host (6 Cores, 12 Threads)" but how much
> memory?
> Thanks,
> E
>
> On Fri, Jan 29, 2016 at 10:00 PM, servtelar <servtelar at gmail.com> wrote:
>
>> Thanks a lot guys for sharing this info. It?s really helpful.
>>
>>
>> On Jan 28, 2016, at 6:18 PM, Sergey Safarov <s.safarov at gmail.com> wrote:
>>
>> We have to core ESXi vm with 140 session (70 calls) with have 70 CPU load.
>>
>> Sergey
>>
>> On Fri, Jan 29, 2016 at 2:01 AM, servtelar <servtelar at gmail.com> wrote:
>>
>>> Hi Chad
>>>
>>> How many legs you are handling with 20 cores on a conference?
>>>
>>> Regards
>>>
>>> Gustavo
>>>
>>> On Jan 28, 2016, at 7:55 PM, Chad Phillips <chad at apartmentlines.com>
>>> wrote:
>>>
>>> I've had very good luck running the newer video branch code on
>>> ProfitBricks: https://www.profitbricks.com/
>>>
>>> As far as I understand, the CPU cycles are guaranteed on their platform.
>>> I've had to put as many as 20 cores on a server to handle some of our
>>> busier video conference calls, but with that it runs quite smoothly.
>>>
>>> On Thu, Jan 28, 2016 at 2:15 PM, Dan Edwards <DEdwards at vertical.com>
>>> wrote:
>>>
>>>> I am reviewing the Confluence Virtualization page and had some
>>>> questions, in particular about VMWare. My company distributes some of its
>>>> software as a VMWare image file and we were looking to distribute a new
>>>> product using FS in the same manner. The products operate at a customer
>>>> premise, on their VMWare infrastructure, not in a cloud environment. Since
>>>> our customers already have VMWare, switching to a different VM
>>>> infrastructure is going to hurt, so I am looking for options/alternatives.
>>>>
>>>> First, does anybody know if the virtual timing issues with VMWare have
>>>> improved since this page was last updated in 2014? Is VMWare still not good
>>>> enough? Is it possible to throw CPU & memory at this and make VMWare good
>>>> enough, or is the virtual timing just not workable?
>>>>
>>>> On the virtualization page, there was a comment from 2010 that you
>>>> might be happy with a High CPU Medium instance on AWS EC2. Certainly
>>>> workload is a factor here, but I am trying to get my head around how big a
>>>> machine to perform how small a workload. Is there a place where people talk
>>>> about their experiences?
>>>>
>>>> Are there other VM platforms that might be acceptable?
>>>>
>>>> Any help or comment is appreciated.
>>>>
>>>> Thank you,
>>>> Dan
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 eschmidbauer at gmail.com  Mon Feb  1 18:38:29 2016
From: eschmidbauer at gmail.com (E. Schmidbauer)
Date: Mon, 1 Feb 2016 10:38:29 -0500
Subject: [Freeswitch-users] FreeSWITCH in virtual environments
In-Reply-To: <CAO_GGAoaFOg8cJ930DHbjyXKxZC1W6Oj7G5KajvjHm=4Gr5FdQ@mail.gmail.com>
References: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C9979B74@PHXEX2.vertical.com>
	<CAJ0sghSaQhKgD4X57DkEGP_MBxHgHi1vTLDh7E7tQXZtKRPz_A@mail.gmail.com>
	<88218521-FB01-4698-9DBF-625D99E8BC79@gmail.com>
	<CAHtxdDfjiAeSEZfnFSoAAK3syGMwBfFrV=Ox+np8AfdTteuqeg@mail.gmail.com>
	<58E2C5C4-87FF-4E01-8048-6C39B9362B02@gmail.com>
	<CAJwe+CBc64_0efV=-0RNr3kB_gtPPU6vTdbcxKWL7m+wGsmbVw@mail.gmail.com>
	<CAO_GGAoaFOg8cJ930DHbjyXKxZC1W6Oj7G5KajvjHm=4Gr5FdQ@mail.gmail.com>
Message-ID: <CAJwe+CCR_fA7i6jMCutEwF8e+c=mEWqV478t_AVB=D3N8Org=A@mail.gmail.com>

Hi Volodymyr,
We are using xml-curl (i dont think that should have too much affect) and
hash counters only.
No database connections, etc...
Thanks-- 24 threads and 8gb seems like a good start!
Emmanuel

On Mon, Feb 1, 2016 at 10:34 AM, Volodymyr Fedorov <lexxua at gmail.com> wrote:

> Hello, it depends.
> In my case 24 threads and 8gb of ram was quite enough. But I used only
> xml-dialplan and hash counters.
>
> On Mon, Feb 1, 2016 at 4:16 PM, E. Schmidbauer <eschmidbauer at gmail.com>
> wrote:
>
>> Can anyone share the VM memory/cpu specs used in these cases?
>> We want to run around 300 CPS on FS (running on vmware).
>> very little transcoding (if any), audio only
>> How much memory/cpu should be provisioned?
>> I see Grant mentioned "single VM host (6 Cores, 12 Threads)" but how
>> much memory?
>> Thanks,
>> E
>>
>> On Fri, Jan 29, 2016 at 10:00 PM, servtelar <servtelar at gmail.com> wrote:
>>
>>> Thanks a lot guys for sharing this info. It?s really helpful.
>>>
>>>
>>> On Jan 28, 2016, at 6:18 PM, Sergey Safarov <s.safarov at gmail.com> wrote:
>>>
>>> We have to core ESXi vm with 140 session (70 calls) with have 70 CPU
>>> load.
>>>
>>> Sergey
>>>
>>> On Fri, Jan 29, 2016 at 2:01 AM, servtelar <servtelar at gmail.com> wrote:
>>>
>>>> Hi Chad
>>>>
>>>> How many legs you are handling with 20 cores on a conference?
>>>>
>>>> Regards
>>>>
>>>> Gustavo
>>>>
>>>> On Jan 28, 2016, at 7:55 PM, Chad Phillips <chad at apartmentlines.com>
>>>> wrote:
>>>>
>>>> I've had very good luck running the newer video branch code on
>>>> ProfitBricks: https://www.profitbricks.com/
>>>>
>>>> As far as I understand, the CPU cycles are guaranteed on their
>>>> platform. I've had to put as many as 20 cores on a server to handle some of
>>>> our busier video conference calls, but with that it runs quite smoothly.
>>>>
>>>> On Thu, Jan 28, 2016 at 2:15 PM, Dan Edwards <DEdwards at vertical.com>
>>>> wrote:
>>>>
>>>>> I am reviewing the Confluence Virtualization page and had some
>>>>> questions, in particular about VMWare. My company distributes some of its
>>>>> software as a VMWare image file and we were looking to distribute a new
>>>>> product using FS in the same manner. The products operate at a customer
>>>>> premise, on their VMWare infrastructure, not in a cloud environment. Since
>>>>> our customers already have VMWare, switching to a different VM
>>>>> infrastructure is going to hurt, so I am looking for options/alternatives.
>>>>>
>>>>> First, does anybody know if the virtual timing issues with VMWare have
>>>>> improved since this page was last updated in 2014? Is VMWare still not good
>>>>> enough? Is it possible to throw CPU & memory at this and make VMWare good
>>>>> enough, or is the virtual timing just not workable?
>>>>>
>>>>> On the virtualization page, there was a comment from 2010 that you
>>>>> might be happy with a High CPU Medium instance on AWS EC2. Certainly
>>>>> workload is a factor here, but I am trying to get my head around how big a
>>>>> machine to perform how small a workload. Is there a place where people talk
>>>>> about their experiences?
>>>>>
>>>>> Are there other VM platforms that might be acceptable?
>>>>>
>>>>> Any help or comment is appreciated.
>>>>>
>>>>> Thank you,
>>>>> Dan
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 brian at freeswitch.org  Mon Feb  1 18:56:20 2016
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Feb 2016 09:56:20 -0600
Subject: [Freeswitch-users] Verto vs. SIP.js
In-Reply-To: <1105687706.20160201163013@seznam.cz>
References: <56AF67C9.2010700@level5.de> <95599933.20160201152539@seznam.cz>
	<003a01d15cfe$d26c3350$774499f0$@botecomm.com>
	<1105687706.20160201163013@seznam.cz>
Message-ID: <CAEJMVkAcYGem8dK4m9Q1nAYqrMxk-RiizKYNACmUQ-0UKTOZ3w@mail.gmail.com>

Thats a knee slapper, When did open source for you to modify considered
vendor lock-in?

On Mon, Feb 1, 2016 at 9:30 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:

> Dobr? den,
>
> i think this Verto its too much VendorLock :)
>
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 pond?l? 1. ?nora 2016, 15:42:48, napsal jste:
> *
>
> Denis, what difficulties did you experience with Verto?
>
> Thanks.
>
> Bote
>
>
> *From:* Denis Jakovlev
> *Sent:* Monday, 01 February, 2016 09:26
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Verto vs. SIP.js
>
> Dobr? den,
>
> I use sip.js. It turned out to be a lot easier that Verto.
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 pond?l? 1. ?nora 2016, 15:12:25, napsal jste:
> > Hi, > I want to setup a Click-2-Call-Button for our website. Is there any
> > significant difference between Verto-Mod and libraries such as SIP.js? >
> I do not really understand the advantage of Verto (if there is any). >
> Thanks in advance, > Thorsten >
> _________________________________________________________________________ *
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


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

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 abaci64 at gmail.com  Mon Feb  1 18:57:34 2016
From: abaci64 at gmail.com (Abaci B)
Date: Mon, 1 Feb 2016 10:57:34 -0500
Subject: [Freeswitch-users] Simple LuaScript to Record Session Failing
In-Reply-To: <FE6AF8FC-E6D4-4011-84E4-DFA526188206@gmail.com>
References: <F47B06F3-9A8D-44ED-8936-7C4C0C9EDC52@gmail.com>
	<CAOiUWxEL3=u3vpqK1TPc6Ws91vkTZ02+BCx0gmyq+pX7e8Kk1Q@mail.gmail.com>
	<CDFC4F0A-FBDF-41AF-8FC4-7B9A188317D1@gmail.com>
	<CAJ0sghQriAOy6QORSpMQPn=-D6p6JxPDWPkSP3tzHfcBcrkdDg@mail.gmail.com>
	<FE6AF8FC-E6D4-4011-84E4-DFA526188206@gmail.com>
Message-ID: <CAFmZFiPM3g1JNhDECN-NedT7iLAmfNaNUj8ujAiB-v6gaerheg@mail.gmail.com>

did you try to answer the new_session before starting the record?

On Mon, Feb 1, 2016 at 10:38 AM, Leonardo Lima Ribeiro <
llribeiro90 at gmail.com> wrote:

> Thanks Chad and Oz.
>
> I tried two things:
>
> 1) Oz suggestion:
>
> local new_session = freeswitch.Session("[origination_caller_id_name='
> 7136694967',origination_caller_id_number='7136694967
> ']sofia/gateway/4_NEXTIVA/3157244022", session);
> new_session:setVariable(?RECORD_STEREO?,?false?)
> new_session:sleep("5000")
> new_session:streamFile("voicemail/vm-goodbye.wav")
> new_session:sleep("10000")
> new_session:hangup()
>
> Did not work?
>
> 2) Chad suggestion:
>
> local new_session = freeswitch.Session("[origination_caller_id_name='
> 7136694967',origination_caller_id_number='7136694967
> ']sofia/gateway/4_NEXTIVA/3157244022", session);
>
> uuid = new_session:get_uuid()
> api = freeswitch.API()
> reply = api:executeString("uuid_record " .. uuid .. " start
> /usr/local/freeswitch/recordings/myrecording.wav")
>
> new_session:sleep("5000")
> new_session:streamFile("voicemail/vm-goodbye.wav")
> new_session:sleep("10000")
> new_session:hangup()
>
> Did not work too?
>
> Both cases I have 5 seconds of silence (first 5000 pause) and then when I
> play the Good Bye wav file it starts to record normally. So I have 5
> seconds of silence, good bye sound and 10 seconds of recording.
>
> Chad, I can?t use session:recordFile because it?s a synchronous command
> that blocks my script.. I need to do a lot of actions like pauses, stream
> files etc., while the call is still going on?
>
> Maybe should I open a ticket for FreeSWITCH support? (I even don?t know if
> it exists hehe, maybe a github issue?)
>
> Thank you,
>
>
> On Jan 30, 2016, at 12:36 AM, Chad Phillips <chad at apartmentlines.com>
> wrote:
>
> have you tried the session method specifically for recording files?
>
>
> https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:recordFile
>
> or if that doesn't work, maybe this:
>
> local new_session = freeswitch.Session("someoriginatestring", session);
> uuid = new_session:get_uuid()
> api = freeswitch.API()
> reply = api:executeString("uuid_record " .. uuid .. " start " .. filepath)
>
> On Fri, Jan 29, 2016 at 1:31 PM, Oz Mortimer <omortimer at gmail.com> wrote:
>
>> since this is a single left call try setting record_stereo to false -
>> https://wiki.freeswitch.org/wiki/Variable_RECORD_STEREO
>> I'm probably wrong, but worth a shot!
>>
>> On 29 Jan 2016, at 19:53, Leonardo Ribeiro <llribeiro90 at gmail.com> wrote:
>>
>> Hello Guys,
>>
>> Any idea?
>> I could not evolve this yet...
>>
>> Thank you,
>>
>> 2016-01-28 18:52 GMT-02:00 Leonardo Lima Ribeiro <llribeiro90 at gmail.com>:
>>
>>> Hello all,
>>>
>>> I?m trying to record an IVR using my gateway to do the outbound call in
>>> my luascript:
>>>
>>> local new_session =
>>> freeswitch.Session("[origination_caller_id_name=?987654321',origination_caller_id_number=?987654321']sofia/gateway/MY_GATEWAY/
>>> 3157244022", session);
>>>
>>> new_session:execute("record_session","/usr/local/freeswitch/recordings/myrecording.wav")
>>> new_session:sleep("10000")
>>> new_session:hangup()
>>>
>>> So in the above script I just call to the Bank Of America as an example
>>> and try to record the first 10 seconds of the call in the recordings path.
>>>
>>> The problem is that I have an empty recording file.. Why?
>>>
>>> The funny thing is: if I add this command after the record_session
>>> command:
>>> new_session:streamFile("voicemail/vm-goodbye.wav?);
>>>
>>> And then this is the entire new script:
>>> local new_session =
>>> freeswitch.Session("[origination_caller_id_name=?987654321',origination_caller_id_number=?987654321']sofia/gateway/MY_GATEWAY/
>>> 3157244022", session);
>>>
>>> new_session:execute("record_session","/usr/local/freeswitch/recordings/myrecording.wav?)
>>> new_session:streamFile("voicemail/vm-goodbye.wav?);
>>> new_session:sleep("10000")
>>> new_session:hangup()
>>>
>>> I can hear the Good Bye sound from my script and then hear the Bank of
>>> America IVR.
>>>
>>> I just don?t understand why the record works if I play a sound in our
>>> side and the record does not work if I don?t play any sound.
>>>
>>> Do you know what?s happening? How can I solve this?
>>>
>>> Thank you!
>>>
>>>
>>
>>
>> --
>> Leonardo Lima Ribeiro
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb  1 19:11:48 2016
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 1 Feb 2016 10:11:48 -0600
Subject: [Freeswitch-users] Verto vs. SIP.js
In-Reply-To: <56AF67C9.2010700@level5.de>
References: <56AF67C9.2010700@level5.de>
Message-ID: <504f01d15d0b$4199d8c0$c4cd8a40$@freeswitch.org>

The difference is,

Sip.js is a full sip stack written in JS, Verto is a lot smaller simpler
stack to use. 

If you don?t need SIP (and really most people really don?t) when you have
WebRTC. Verto greatly simplifies the calling. 


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Thorsten
G?llner
Sent: Monday, February 1, 2016 8:12 AM
To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Verto vs. SIP.js

Hi,

I want to setup a Click-2-Call-Button for our website. Is there any
significant difference between Verto-Mod and libraries such as SIP.js?

I do not really understand the advantage of Verto (if there is any).

Thanks in advance,

Thorsten

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

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

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



From krice at freeswitch.org  Mon Feb  1 19:12:43 2016
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 1 Feb 2016 10:12:43 -0600
Subject: [Freeswitch-users] Verto vs. SIP.js
In-Reply-To: <1105687706.20160201163013@seznam.cz>
References: <56AF67C9.2010700@level5.de> <95599933.20160201152539@seznam.cz>
	<003a01d15cfe$d26c3350$774499f0$@botecomm.com>
	<1105687706.20160201163013@seznam.cz>
Message-ID: <505001d15d0b$623021c0$26906540$@freeswitch.org>

Really? That's why the whole protocol is OpenSource. not much vendor lock in
there. it can be used anywhere. 

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Denis
Jakovlev
Sent: Monday, February 1, 2016 9:30 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Verto vs. SIP.js

 

Dobr? den,

i think this Verto its too much VendorLock :) 

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

pond?l? 1. ?nora 2016, 15:42:48, napsal jste:

	
Denis, what difficulties did you experience with Verto?

Thanks.

Bote


From: Denis Jakovlev
Sent: Monday, 01 February, 2016 09:26
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Verto vs. SIP.js

Dobr? den,

I use sip.js. It turned out to be a lot easier that Verto.

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

pond?l? 1. ?nora 2016, 15:12:25, napsal jste:

> Hi,

> I want to setup a Click-2-Call-Button for our website. Is there any 
> significant difference between Verto-Mod and libraries such as SIP.js?

> I do not really understand the advantage of Verto (if there is any).

> Thanks in advance,

> Thorsten

> _________________________________________________________________________

 

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From giacomo.vacca at gmail.com  Mon Feb  1 19:22:24 2016
From: giacomo.vacca at gmail.com (Giacomo Vacca)
Date: Mon, 1 Feb 2016 17:22:24 +0100
Subject: [Freeswitch-users] Verto vs. SIP.js
In-Reply-To: <505001d15d0b$623021c0$26906540$@freeswitch.org>
References: <56AF67C9.2010700@level5.de> <95599933.20160201152539@seznam.cz>
	<003a01d15cfe$d26c3350$774499f0$@botecomm.com>
	<1105687706.20160201163013@seznam.cz>
	<505001d15d0b$623021c0$26906540$@freeswitch.org>
Message-ID: <CAO6UKa2Tm5j2hF5fCnmNpgxT=7G9mpx5jagw_7PyYWAKTXQAGw@mail.gmail.com>

With Verto you can also be more creative, see for example:
https://freeswitch.org/verto-not-just-for-call-signaling/
It also provides an interesting feature: "attach". e.g. if a tab is closed
by mistake, FS can resume automatically the session upon reconnection (with
the same ID).
Also being a simple JSON-based protocol you can have people working on the
client side even with limited knowledge of SIP.

On 1 February 2016 at 17:12, Ken Rice <krice at freeswitch.org> wrote:

> Really? That?s why the whole protocol is OpenSource? not much vendor lock
> in there? it can be used anywhere?
>
>
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Denis
> Jakovlev
> *Sent:* Monday, February 1, 2016 9:30 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Verto vs. SIP.js
>
>
>
> Dobr? den,
>
> i think this Verto its too much VendorLock :)
>
>
>
>
>
>
> *-- S pozdravem,Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382pond?l? 1. ?nora 2016, 15:42:48, napsal jste:*
>
> Denis, what difficulties did you experience with Verto?
>
> Thanks.
>
> Bote
>
>
> *From:* Denis Jakovlev
> *Sent:* Monday, 01 February, 2016 09:26
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Verto vs. SIP.js
>
> Dobr? den,
>
> I use sip.js. It turned out to be a lot easier that Verto.
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> *-- S pozdravem,Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382pond?l? 1. ?nora 2016, 15:12:25, napsal jste:>
> Hi,> I want to setup a Click-2-Call-Button for our website. Is there any >
> significant difference between Verto-Mod and libraries such as SIP.js?> I
> do not really understand the advantage of Verto (if there is any).> Thanks
> in advance,> Thorsten>
> _________________________________________________________________________*
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bote_radio at botecomm.com  Mon Feb  1 19:26:28 2016
From: bote_radio at botecomm.com (Bote Man)
Date: Mon, 1 Feb 2016 11:26:28 -0500
Subject: [Freeswitch-users] Verto vs. SIP.js
In-Reply-To: <505001d15d0b$623021c0$26906540$@freeswitch.org>
References: <56AF67C9.2010700@level5.de>
	<95599933.20160201152539@seznam.cz>	<003a01d15cfe$d26c3350$774499f0$@botecomm.com>	<1105687706.20160201163013@seznam.cz>
	<505001d15d0b$623021c0$26906540$@freeswitch.org>
Message-ID: <007201d15d0d$4d80cde0$e88269a0$@botecomm.com>

Perhaps Denis should see what can be done with Verto on the weekly
conference call?

 

https://cantina.freeswitch.org/vc

 

Enjoy.

 

Bote

 

 

From: Ken Rice
Sent: Monday, 01 February, 2016 11:13
Subject: Re: [Freeswitch-users] Verto vs. SIP.js

 

Really? That's why the whole protocol is OpenSource. not much vendor lock in
there. it can be used anywhere. 

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Denis
Jakovlev
Sent: Monday, February 1, 2016 9:30 AM
Subject: Re: [Freeswitch-users] Verto vs. SIP.js

 

Dobr? den,

i think this Verto its too much VendorLock :) 

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

pond?l? 1. ?nora 2016, 15:42:48, napsal jste:

	
Denis, what difficulties did you experience with Verto?

Thanks.

Bote


From: Denis Jakovlev
Sent: Monday, 01 February, 2016 09:26
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Verto vs. SIP.js

Dobr? den,

I use sip.js. It turned out to be a lot easier that Verto.

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

pond?l? 1. ?nora 2016, 15:12:25, napsal jste:

> Hi,

> I want to setup a Click-2-Call-Button for our website. Is there any 
> significant difference between Verto-Mod and libraries such as SIP.js?

> I do not really understand the advantage of Verto (if there is any).

> Thanks in advance,

> Thorsten

> _________________________________________________________________________

 

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From bote_radio at botecomm.com  Mon Feb  1 19:26:28 2016
From: bote_radio at botecomm.com (Bote Man)
Date: Mon, 1 Feb 2016 11:26:28 -0500
Subject: [Freeswitch-users] FreeSWITCH in virtual environments
In-Reply-To: <CAJwe+CCR_fA7i6jMCutEwF8e+c=mEWqV478t_AVB=D3N8Org=A@mail.gmail.com>
References: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C9979B74@PHXEX2.vertical.com>	<CAJ0sghSaQhKgD4X57DkEGP_MBxHgHi1vTLDh7E7tQXZtKRPz_A@mail.gmail.com>	<88218521-FB01-4698-9DBF-625D99E8BC79@gmail.com>	<CAHtxdDfjiAeSEZfnFSoAAK3syGMwBfFrV=Ox+np8AfdTteuqeg@mail.gmail.com>	<58E2C5C4-87FF-4E01-8048-6C39B9362B02@gmail.com>	<CAJwe+CBc64_0efV=-0RNr3kB_gtPPU6vTdbcxKWL7m+wGsmbVw@mail.gmail.com>	<CAO_GGAoaFOg8cJ930DHbjyXKxZC1W6Oj7G5KajvjHm=4Gr5FdQ@mail.gmail.com>
	<CAJwe+CCR_fA7i6jMCutEwF8e+c=mEWqV478t_AVB=D3N8Org=A@mail.gmail.com>
Message-ID: <007701d15d0d$4dfba330$e9f2e990$@botecomm.com>

It seems to me that the XML dialplan routing process would consume more cpu than a short and simple curl request to a database server, no?

 

Anyway, there are probably many things that can be done to streamline performance for a particular application. I know on Windoze it was once suggested to me a long time ago to disable the operating system updates of file modification time stamps, for example. That can add up when a busy system is writing to log files and creating individual CDR files to be read and deleted by an accounting process. I don?t know how this would apply to linux, but it demonstrates how far out of the box one can look for performance improvements.

 

In any case, when discussing virtualization performance it is essential to provide specifics of the instance that runs FreeSWITCH. If someone reports that FS ran very poorly, but does not say that it was a tiny instance that was starved for resources then we can?t evaluate that report fairly. 

 

So the specifics in the report by Volodymyr are useful. Please keep them coming! I am compiling these anecdotes on Confluence for others to read in the future.

 

Thanks.

 

Bote

 

 

From: E. Schmidbauer
Sent: Monday, 01 February, 2016 10:38
Subject: Re: [Freeswitch-users] FreeSWITCH in virtual environments

 

Hi Volodymyr,

We are using xml-curl (i dont think that should have too much affect) and hash counters only.

No database connections, etc... 

Thanks-- 24 threads and 8gb seems like a good start!

Emmanuel

 

On Mon, Feb 1, 2016 at 10:34 AM, Volodymyr Fedorov <lexxua at gmail.com> wrote:

Hello, it depends.

In my case 24 threads and 8gb of ram was quite enough. But I used only xml-dialplan and hash counters.

 

On Mon, Feb 1, 2016 at 4:16 PM, E. Schmidbauer <eschmidbauer at gmail.com> wrote:

Can anyone share the VM memory/cpu specs used in these cases?
We want to run around 300 CPS on FS (running on vmware).

very little transcoding (if any), audio only

How much memory/cpu should be provisioned? 
I see Grant mentioned "single VM host (6 Cores, 12 Threads)" but how much memory?

Thanks,

E

 

On Fri, Jan 29, 2016 at 10:00 PM, servtelar <servtelar at gmail.com> wrote:

Thanks a lot guys for sharing this info. It?s really helpful.

 

 

On Jan 28, 2016, at 6:18 PM, Sergey Safarov <s.safarov at gmail.com> wrote:

 

We have to core ESXi vm with 140 session (70 calls) with have 70 CPU load.

 

Sergey

 

On Fri, Jan 29, 2016 at 2:01 AM, servtelar <servtelar at gmail.com> wrote:

Hi Chad

 

How many legs you are handling with 20 cores on a conference?

 

Regards

 

Gustavo

 

On Jan 28, 2016, at 7:55 PM, Chad Phillips <chad at apartmentlines.com> wrote:

 

I've had very good luck running the newer video branch code on ProfitBricks: https://www.profitbricks.com/

 

As far as I understand, the CPU cycles are guaranteed on their platform. I've had to put as many as 20 cores on a server to handle some of our busier video conference calls, but with that it runs quite smoothly.

 

On Thu, Jan 28, 2016 at 2:15 PM, Dan Edwards <DEdwards at vertical.com> wrote:

I am reviewing the Confluence Virtualization page and had some questions, in particular about VMWare. My company distributes some of its software as a VMWare image file and we were looking to distribute a new product using FS in the same manner. The products operate at a customer premise, on their VMWare infrastructure, not in a cloud environment. Since our customers already have VMWare, switching to a different VM infrastructure is going to hurt, so I am looking for options/alternatives.

First, does anybody know if the virtual timing issues with VMWare have improved since this page was last updated in 2014? Is VMWare still not good enough? Is it possible to throw CPU & memory at this and make VMWare good enough, or is the virtual timing just not workable?

On the virtualization page, there was a comment from 2010 that you might be happy with a High CPU Medium instance on AWS EC2. Certainly workload is a factor here, but I am trying to get my head around how big a machine to perform how small a workload. Is there a place where people talk about their experiences?

Are there other VM platforms that might be acceptable?

Any help or comment is appreciated.

Thank you,
Dan


_________________________________________________________________________



 

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From llribeiro90 at gmail.com  Mon Feb  1 19:50:16 2016
From: llribeiro90 at gmail.com (Leonardo Lima Ribeiro)
Date: Mon, 1 Feb 2016 14:50:16 -0200
Subject: [Freeswitch-users] Simple LuaScript to Record Session Failing
In-Reply-To: <CAFmZFiPM3g1JNhDECN-NedT7iLAmfNaNUj8ujAiB-v6gaerheg@mail.gmail.com>
References: <F47B06F3-9A8D-44ED-8936-7C4C0C9EDC52@gmail.com>
	<CAOiUWxEL3=u3vpqK1TPc6Ws91vkTZ02+BCx0gmyq+pX7e8Kk1Q@mail.gmail.com>
	<CDFC4F0A-FBDF-41AF-8FC4-7B9A188317D1@gmail.com>
	<CAJ0sghQriAOy6QORSpMQPn=-D6p6JxPDWPkSP3tzHfcBcrkdDg@mail.gmail.com>
	<FE6AF8FC-E6D4-4011-84E4-DFA526188206@gmail.com>
	<CAFmZFiPM3g1JNhDECN-NedT7iLAmfNaNUj8ujAiB-v6gaerheg@mail.gmail.com>
Message-ID: <F93D8FB2-3148-4A44-AFA9-FFA25626CB98@gmail.com>

Answer in what extension? I?m just calling to an outbound IVRs and doing some stuffs like sending dtmfs, playing sounds etc. to interact with them.

I?m not using any extensions for this type of calls? Is this really necessary?

Thank you,

> On Feb 1, 2016, at 1:57 PM, Abaci B <abaci64 at gmail.com> wrote:
> 
> did you try to answer the new_session before starting the record?
> 
> On Mon, Feb 1, 2016 at 10:38 AM, Leonardo Lima Ribeiro <llribeiro90 at gmail.com <mailto:llribeiro90 at gmail.com>> wrote:
> Thanks Chad and Oz.
> 
> I tried two things:
> 
> 1) Oz suggestion:
> 
> local new_session = freeswitch.Session("[origination_caller_id_name='7136694967 <tel:7136694967>',origination_caller_id_number='7136694967 <tel:7136694967>']sofia/gateway/4_NEXTIVA/3157244022 <tel:3157244022>", session);
> new_session:setVariable(?RECORD_STEREO?,?false?)
> new_session:sleep("5000")
> new_session:streamFile("voicemail/vm-goodbye.wav")
> new_session:sleep("10000")
> new_session:hangup()
> 
> Did not work?
> 
> 2) Chad suggestion:
> 
> local new_session = freeswitch.Session("[origination_caller_id_name='7136694967 <tel:7136694967>',origination_caller_id_number='7136694967 <tel:7136694967>']sofia/gateway/4_NEXTIVA/3157244022 <tel:3157244022>", session);
> 
> uuid = new_session:get_uuid()
> api = freeswitch.API()
> reply = api:executeString("uuid_record " .. uuid .. " start /usr/local/freeswitch/recordings/myrecording.wav")
> 
> new_session:sleep("5000")
> new_session:streamFile("voicemail/vm-goodbye.wav")
> new_session:sleep("10000")
> new_session:hangup()
> 
> Did not work too? 
> 
> Both cases I have 5 seconds of silence (first 5000 pause) and then when I play the Good Bye wav file it starts to record normally. So I have 5 seconds of silence, good bye sound and 10 seconds of recording.
> 
> Chad, I can?t use session:recordFile because it?s a synchronous command that blocks my script.. I need to do a lot of actions like pauses, stream files etc., while the call is still going on?
> 
> Maybe should I open a ticket for FreeSWITCH support? (I even don?t know if it exists hehe, maybe a github issue?)
> 
> Thank you,
> 
> 
>> On Jan 30, 2016, at 12:36 AM, Chad Phillips <chad at apartmentlines.com <mailto:chad at apartmentlines.com>> wrote:
>> 
>> have you tried the session method specifically for recording files?
>> 
>> https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:recordFile <https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:recordFile>
>> 
>> or if that doesn't work, maybe this:
>> 
>> local new_session = freeswitch.Session("someoriginatestring", session);
>> uuid = new_session:get_uuid()
>> api = freeswitch.API()
>> reply = api:executeString("uuid_record " .. uuid .. " start " .. filepath)
>> 
>> On Fri, Jan 29, 2016 at 1:31 PM, Oz Mortimer <omortimer at gmail.com <mailto:omortimer at gmail.com>> wrote:
>> since this is a single left call try setting record_stereo to false - https://wiki.freeswitch.org/wiki/Variable_RECORD_STEREO <https://wiki.freeswitch.org/wiki/Variable_RECORD_STEREO>
>> I'm probably wrong, but worth a shot!
>> 
>> On 29 Jan 2016, at 19:53, Leonardo Ribeiro <llribeiro90 at gmail.com <mailto:llribeiro90 at gmail.com>> wrote:
>> 
>>> Hello Guys,
>>> 
>>> Any idea?
>>> I could not evolve this yet...
>>> 
>>> Thank you,
>>> 
>>> 2016-01-28 18:52 GMT-02:00 Leonardo Lima Ribeiro <llribeiro90 at gmail.com <mailto:llribeiro90 at gmail.com>>:
>>> Hello all,
>>> 
>>> I?m trying to record an IVR using my gateway to do the outbound call in my luascript:
>>> 
>>> local new_session = freeswitch.Session("[origination_caller_id_name=?987654321',origination_caller_id_number=?987654321']sofia/gateway/MY_GATEWAY/3157244022 <tel:3157244022>", session);
>>> new_session:execute("record_session","/usr/local/freeswitch/recordings/myrecording.wav")
>>> new_session:sleep("10000")
>>> new_session:hangup()
>>> 
>>> So in the above script I just call to the Bank Of America as an example and try to record the first 10 seconds of the call in the recordings path.
>>> 
>>> The problem is that I have an empty recording file.. Why?
>>> 
>>> The funny thing is: if I add this command after the record_session command: 
>>> new_session:streamFile("voicemail/vm-goodbye.wav?);
>>> 
>>> And then this is the entire new script:
>>> local new_session = freeswitch.Session("[origination_caller_id_name=?987654321',origination_caller_id_number=?987654321']sofia/gateway/MY_GATEWAY/3157244022 <tel:3157244022>", session);
>>> new_session:execute("record_session","/usr/local/freeswitch/recordings/myrecording.wav?)
>>> new_session:streamFile("voicemail/vm-goodbye.wav?);
>>> new_session:sleep("10000")
>>> new_session:hangup()
>>> 
>>> I can hear the Good Bye sound from my script and then hear the Bank of America IVR.
>>> 
>>> I just don?t understand why the record works if I play a sound in our side and the record does not work if I don?t play any sound.
>>> 
>>> Do you know what?s happening? How can I solve this?
>>> 
>>> Thank you!
>>> 
>>> 
>>> 
>>> 
>>> -- 
>>> Leonardo Lima Ribeiro
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services: 
>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>>> http://www.cluecon.com <http://www.cluecon.com/>
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: 
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 chad at apartmentlines.com  Mon Feb  1 19:46:01 2016
From: chad at apartmentlines.com (Chad Phillips)
Date: Mon, 1 Feb 2016 09:46:01 -0700
Subject: [Freeswitch-users] FreeSWITCH in virtual environments
In-Reply-To: <007701d15d0d$4dfba330$e9f2e990$@botecomm.com>
References: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C9979B74@PHXEX2.vertical.com>
	<CAJ0sghSaQhKgD4X57DkEGP_MBxHgHi1vTLDh7E7tQXZtKRPz_A@mail.gmail.com>
	<88218521-FB01-4698-9DBF-625D99E8BC79@gmail.com>
	<CAHtxdDfjiAeSEZfnFSoAAK3syGMwBfFrV=Ox+np8AfdTteuqeg@mail.gmail.com>
	<58E2C5C4-87FF-4E01-8048-6C39B9362B02@gmail.com>
	<CAJwe+CBc64_0efV=-0RNr3kB_gtPPU6vTdbcxKWL7m+wGsmbVw@mail.gmail.com>
	<CAO_GGAoaFOg8cJ930DHbjyXKxZC1W6Oj7G5KajvjHm=4Gr5FdQ@mail.gmail.com>
	<CAJwe+CCR_fA7i6jMCutEwF8e+c=mEWqV478t_AVB=D3N8Org=A@mail.gmail.com>
	<007701d15d0d$4dfba330$e9f2e990$@botecomm.com>
Message-ID: <CAJ0sghRyA8YhuMiJGN=x9iV_sh1Q=kkoHzTkeUaAZ_XZaEKbWQ@mail.gmail.com>

In my profitbricks/videoconference scenario, I've been going with 10GB
memory, and that's been fine. i might even be able to get away with less.
you can probably find the CPU specs on profitbricks website, or certainly
by contacting them, i don't know them offhand.

On Mon, Feb 1, 2016 at 9:26 AM, Bote Man <bote_radio at botecomm.com> wrote:

> It seems to me that the XML dialplan routing process would consume more
> cpu than a short and simple curl request to a database server, no?
>
>
>
> Anyway, there are probably many things that can be done to streamline
> performance for a particular application. I know on Windoze it was once
> suggested to me a long time ago to disable the operating system updates of
> file modification time stamps, for example. That can add up when a busy
> system is writing to log files and creating individual CDR files to be read
> and deleted by an accounting process. I don?t know how this would apply to
> linux, but it demonstrates how far out of the box one can look for
> performance improvements.
>
>
>
> In any case, when discussing virtualization performance it is essential to
> provide specifics of the instance that runs FreeSWITCH. If someone reports
> that FS ran very poorly, but does not say that it was a tiny instance that
> was starved for resources then we can?t evaluate that report fairly.
>
>
>
> So the specifics in the report by Volodymyr are useful. Please keep them
> coming! I am compiling these anecdotes on Confluence for others to read in
> the future.
>
>
>
> Thanks.
>
>
>
> Bote
>
>
>
>
>
> *From:* E. Schmidbauer
> *Sent:* Monday, 01 February, 2016 10:38
> *Subject:* Re: [Freeswitch-users] FreeSWITCH in virtual environments
>
>
>
> Hi Volodymyr,
>
> We are using xml-curl (i dont think that should have too much affect) and
> hash counters only.
>
> No database connections, etc...
>
> Thanks-- 24 threads and 8gb seems like a good start!
>
> Emmanuel
>
>
>
> On Mon, Feb 1, 2016 at 10:34 AM, Volodymyr Fedorov <lexxua at gmail.com>
> wrote:
>
> Hello, it depends.
>
> In my case 24 threads and 8gb of ram was quite enough. But I used only
> xml-dialplan and hash counters.
>
>
>
> On Mon, Feb 1, 2016 at 4:16 PM, E. Schmidbauer <eschmidbauer at gmail.com>
> wrote:
>
> Can anyone share the VM memory/cpu specs used in these cases?
> We want to run around 300 CPS on FS (running on vmware).
>
> very little transcoding (if any), audio only
>
> How much memory/cpu should be provisioned?
> I see Grant mentioned "single VM host (6 Cores, 12 Threads)" but how much
> memory?
>
> Thanks,
>
> E
>
>
>
> On Fri, Jan 29, 2016 at 10:00 PM, servtelar <servtelar at gmail.com> wrote:
>
> Thanks a lot guys for sharing this info. It?s really helpful.
>
>
>
>
>
> On Jan 28, 2016, at 6:18 PM, Sergey Safarov <s.safarov at gmail.com> wrote:
>
>
>
> We have to core ESXi vm with 140 session (70 calls) with have 70 CPU load.
>
>
>
> Sergey
>
>
>
> On Fri, Jan 29, 2016 at 2:01 AM, servtelar <servtelar at gmail.com> wrote:
>
> Hi Chad
>
>
>
> How many legs you are handling with 20 cores on a conference?
>
>
>
> Regards
>
>
>
> Gustavo
>
>
>
> On Jan 28, 2016, at 7:55 PM, Chad Phillips <chad at apartmentlines.com>
> wrote:
>
>
>
> I've had very good luck running the newer video branch code on
> ProfitBricks: https://www.profitbricks.com/
>
>
>
> As far as I understand, the CPU cycles are guaranteed on their platform.
> I've had to put as many as 20 cores on a server to handle some of our
> busier video conference calls, but with that it runs quite smoothly.
>
>
>
> On Thu, Jan 28, 2016 at 2:15 PM, Dan Edwards <DEdwards at vertical.com>
> wrote:
>
> I am reviewing the Confluence Virtualization page and had some questions,
> in particular about VMWare. My company distributes some of its software as
> a VMWare image file and we were looking to distribute a new product using
> FS in the same manner. The products operate at a customer premise, on their
> VMWare infrastructure, not in a cloud environment. Since our customers
> already have VMWare, switching to a different VM infrastructure is going to
> hurt, so I am looking for options/alternatives.
>
> First, does anybody know if the virtual timing issues with VMWare have
> improved since this page was last updated in 2014? Is VMWare still not good
> enough? Is it possible to throw CPU & memory at this and make VMWare good
> enough, or is the virtual timing just not workable?
>
> On the virtualization page, there was a comment from 2010 that you might
> be happy with a High CPU Medium instance on AWS EC2. Certainly workload is
> a factor here, but I am trying to get my head around how big a machine to
> perform how small a workload. Is there a place where people talk about
> their experiences?
>
> Are there other VM platforms that might be acceptable?
>
> Any help or comment is appreciated.
>
> Thank you,
> Dan
>
>
> _________________________________________________________________________
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 diriver63 at gmail.com  Tue Feb  2 02:09:16 2016
From: diriver63 at gmail.com (Diego Rivera)
Date: Mon, 1 Feb 2016 17:09:16 -0600
Subject: [Freeswitch-users] T.38 US providers
Message-ID: <CAOqKEiuM0xMS-SfzY7gh-eretM+5wjOR8YyT00635JRBqAC9Lw@mail.gmail.com>

Hello all,
We use Freeswitch as our main fax server with a T38 trunk from Babytel, but
now we're looking for alternatives on other reliable T38 providers in the
US... mostly to see if we can get a comparable success rate at a better
price.

We typically fax in and out about 250k pages a week... What other good T.38
trunk providers do you guys know/recommend?


Thanks,

Diego
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From gregor at infomedia.si  Tue Feb  2 01:37:23 2016
From: gregor at infomedia.si (Gregor)
Date: Mon, 1 Feb 2016 22:37:23 +0000 (UTC)
Subject: [Freeswitch-users] TCP registrations
Message-ID: <loom.20160201T233412-829@post.gmane.org>

I think I am missing something.

I would like to configure freeswitch that listens on TCP port for client 
registrations (internal profile). As I read, freeswitch should do this by 
default. But freeswitch responses only on UDP protocol. Is there a conf 
setting for specify also tcp for registrations.



From sandeep.goje at gmail.com  Mon Feb  1 14:14:42 2016
From: sandeep.goje at gmail.com (sandeep goje)
Date: Mon, 1 Feb 2016 16:44:42 +0530
Subject: [Freeswitch-users] Sending Data in SIP BYE message
Message-ID: <CAN9cf2UH8LfNVpp1Q1_G9oLwxMs36cnaSkyi1+8BrEL9Grg6cA@mail.gmail.com>

Hi,
I have a issue while setting the sip_bye_h_ headers. Here is the scenario
A-->B

A is a sip end point and B is a PSTN call placed through the dialplan.

In this scenario, when B hangs up the call, I get proper Bye message sent
to sip_bye_h headers set in the Bye message.But when A hangs up, the
sip_bye_h headers are not sent in the 200 OK message. Is there a way to
send the sip_bye_h headers in both BYE and 200 OK messages.

I am trying to send bridge_channel or channel_name from B to A.

Regards,

Sandeep


The most profound statements are often said in silence.
-Lynn Johnston
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From bote_radio at botecomm.com  Tue Feb  2 03:32:59 2016
From: bote_radio at botecomm.com (Bote Man)
Date: Mon, 1 Feb 2016 19:32:59 -0500
Subject: [Freeswitch-users] TCP registrations
In-Reply-To: <loom.20160201T233412-829@post.gmane.org>
References: <loom.20160201T233412-829@post.gmane.org>
Message-ID: <001b01d15d51$455900d0$d00b0270$@botecomm.com>

FreeSWITCH uses UDP by default for SIP signaling. You can change this in the
SIP_profile I believe.


---
Bote

FreeSWITCH Docs Janitor
http://freeswitch.org/confluence




> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> users-bounces at lists.freeswitch.org] On Behalf Of Gregor
> Sent: Monday, 01 February, 2016 17:37
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] TCP registrations
> 
> I think I am missing something.
> 
> I would like to configure freeswitch that listens on TCP port for client
> registrations (internal profile). As I read, freeswitch should do this by
> default. But freeswitch responses only on UDP protocol. Is there a conf
> setting for specify also tcp for registrations.
> 
> 
> __________________________________________________________



From s.safarov at gmail.com  Tue Feb  2 06:07:52 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 02 Feb 2016 03:07:52 +0000
Subject: [Freeswitch-users] recording problems with mod_shout
	debian/master
In-Reply-To: <CAEvFzR8Xxu57=rLnbwPkMJFAGMJXn7O2cik41YuFv128BcwZ9g@mail.gmail.com>
References: <CAEvFzR8Xxu57=rLnbwPkMJFAGMJXn7O2cik41YuFv128BcwZ9g@mail.gmail.com>
Message-ID: <CAHtxdDcnCL5LrL0yG=rM=G-bqCWP2TFQWBMpN37uXaQ4w6QYKA@mail.gmail.com>

I has same case FS-8686.

On Mon, Feb 1, 2016, 16:49 Luis Azedo <luis.azedo at factorlusitano.com> wrote:

> Hi,
>
> anyone having problems recording in mp3  format ?
>
> 2016-02-01 13:41:47.108078 [INFO] kazoo_node.c:625 exec: *uuid_record(96722133-5060-508 at BJC.BGI.CG.BD
> <96722133-5060-508 at BJC.BGI.CG.BD> start
> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3 600)*
> 2016-02-01 13:41:47.108078 [DEBUG]* switch_core_file.c:323 File
> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3 sample rate 8000 doesn't match
> requested rate 16000*
> 2016-02-01 13:41:47.128429 [DEBUG] *switch_core_media_bug.c:828 Attaching
> BUG to sofia/sipinterface_1/995582142 at teste.sip.90e9.com
> <995582142 at teste.sip.90e9.com>*
> 2016-02-01 13:41:47.148020 [DEBUG] *switch_ivr_async.c:1491 No silence
> detection configured; assuming start of speech*
> 2016-02-01 13:41:47.168888 [INFO] *mod_shout.c:326 LAME 3.99.5 64bits
> (http://lame.sf.net <http://lame.sf.net>)*
> 2016-02-01 13:41:47.168888 [INFO] *mod_shout.c:326 polyphase lowpass
> filter disabled*
> 2016-02-01 13:41:47.168888 [ERR] *mod_shout.c:1057 MP3 encode error -1!*
> 2016-02-01 13:41:47.168888 [ERR] *switch_ivr_async.c:1160 Error writing
> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3*
> 2016-02-01 13:41:47.168888 [ERR] *mod_shout.c:1057 MP3 encode error -1!*
> 2016-02-01 13:41:47.168888 [ERR] *switch_ivr_async.c:1160 Error writing
> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3*
> 2016-02-01 13:41:47.688430 [ERR] *mod_shout.c:1057 MP3 encode error -1!*
> 2016-02-01 13:41:47.688430 [ERR] *switch_ivr_async.c:1160 Error writing
> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3*
> 2016-02-01 13:41:47.688430 [ERR] *mod_shout.c:1057 MP3 encode error -1!*
> 2016-02-01 13:41:47.688430 [ERR] *switch_ivr_async.c:1160 Error writing
> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3*
> 2016-02-01 13:41:47.688430 [ERR] *mod_shout.c:1057 MP3 encode error -1!*
> 2016-02-01 13:41:47.688430 [ERR] *switch_ivr_async.c:1160 Error writing
> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3*
>
> 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 s.safarov at gmail.com  Tue Feb  2 06:57:16 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 02 Feb 2016 03:57:16 +0000
Subject: [Freeswitch-users] recording problems with mod_shout
	debian/master
In-Reply-To: <CAHtxdDcnCL5LrL0yG=rM=G-bqCWP2TFQWBMpN37uXaQ4w6QYKA@mail.gmail.com>
References: <CAEvFzR8Xxu57=rLnbwPkMJFAGMJXn7O2cik41YuFv128BcwZ9g@mail.gmail.com>
	<CAHtxdDcnCL5LrL0yG=rM=G-bqCWP2TFQWBMpN37uXaQ4w6QYKA@mail.gmail.com>
Message-ID: <CAHtxdDfyWQUWE2iCyGsjgz2tWf9fSgrm1q9DbSz3HtaCt590Ew@mail.gmail.com>

Luisu you has additional info of bug.
Could you reproduce this on testbox and made core dump?
For me is interested core at switch_core_file.c:323

On Tue, Feb 2, 2016, 06:07 Sergey Safarov <s.safarov at gmail.com> wrote:

> I has same case FS-8686.
>
> On Mon, Feb 1, 2016, 16:49 Luis Azedo <luis.azedo at factorlusitano.com>
> wrote:
>
>> Hi,
>>
>> anyone having problems recording in mp3  format ?
>>
>> 2016-02-01 13:41:47.108078 [INFO] kazoo_node.c:625 exec: *uuid_record(96722133-5060-508 at BJC.BGI.CG.BD
>> <96722133-5060-508 at BJC.BGI.CG.BD> start
>> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3 600)*
>> 2016-02-01 13:41:47.108078 [DEBUG]* switch_core_file.c:323 File
>> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3 sample rate 8000 doesn't match
>> requested rate 16000*
>> 2016-02-01 13:41:47.128429 [DEBUG] *switch_core_media_bug.c:828
>> Attaching BUG to sofia/sipinterface_1/995582142 at teste.sip.90e9.com
>> <995582142 at teste.sip.90e9.com>*
>> 2016-02-01 13:41:47.148020 [DEBUG] *switch_ivr_async.c:1491 No silence
>> detection configured; assuming start of speech*
>> 2016-02-01 13:41:47.168888 [INFO] *mod_shout.c:326 LAME 3.99.5 64bits
>> (http://lame.sf.net <http://lame.sf.net>)*
>> 2016-02-01 13:41:47.168888 [INFO] *mod_shout.c:326 polyphase lowpass
>> filter disabled*
>> 2016-02-01 13:41:47.168888 [ERR] *mod_shout.c:1057 MP3 encode error -1!*
>> 2016-02-01 13:41:47.168888 [ERR] *switch_ivr_async.c:1160 Error writing
>> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3*
>> 2016-02-01 13:41:47.168888 [ERR] *mod_shout.c:1057 MP3 encode error -1!*
>> 2016-02-01 13:41:47.168888 [ERR] *switch_ivr_async.c:1160 Error writing
>> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3*
>> 2016-02-01 13:41:47.688430 [ERR] *mod_shout.c:1057 MP3 encode error -1!*
>> 2016-02-01 13:41:47.688430 [ERR] *switch_ivr_async.c:1160 Error writing
>> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3*
>> 2016-02-01 13:41:47.688430 [ERR] *mod_shout.c:1057 MP3 encode error -1!*
>> 2016-02-01 13:41:47.688430 [ERR] *switch_ivr_async.c:1160 Error writing
>> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3*
>> 2016-02-01 13:41:47.688430 [ERR] *mod_shout.c:1057 MP3 encode error -1!*
>> 2016-02-01 13:41:47.688430 [ERR] *switch_ivr_async.c:1160 Error writing
>> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3*
>>
>> 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 denis at ringme.ru  Tue Feb  2 11:43:25 2016
From: denis at ringme.ru (=?UTF-8?B?0JTQtdC90LjRgQ==?=)
Date: Tue, 2 Feb 2016 11:43:25 +0300
Subject: [Freeswitch-users] notes about amazon AWS?
Message-ID: <56B06C2D.5010505@ringme.ru>

Hello, who can share AMI for freeswitch and maybe some notes about setup 
(best practices)? But need centos 6 x86-64, it's _requirement_ some 
other our software.

PV or HWM mode?
https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 - too 
few info about centos.


From ssinyagin at gmail.com  Tue Feb  2 12:06:31 2016
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Tue, 2 Feb 2016 10:06:31 +0100
Subject: [Freeswitch-users] FreeSWITCH in virtual environments
In-Reply-To: <CAJ0sghRyA8YhuMiJGN=x9iV_sh1Q=kkoHzTkeUaAZ_XZaEKbWQ@mail.gmail.com>
References: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C9979B74@PHXEX2.vertical.com>
	<CAJ0sghSaQhKgD4X57DkEGP_MBxHgHi1vTLDh7E7tQXZtKRPz_A@mail.gmail.com>
	<88218521-FB01-4698-9DBF-625D99E8BC79@gmail.com>
	<CAHtxdDfjiAeSEZfnFSoAAK3syGMwBfFrV=Ox+np8AfdTteuqeg@mail.gmail.com>
	<58E2C5C4-87FF-4E01-8048-6C39B9362B02@gmail.com>
	<CAJwe+CBc64_0efV=-0RNr3kB_gtPPU6vTdbcxKWL7m+wGsmbVw@mail.gmail.com>
	<CAO_GGAoaFOg8cJ930DHbjyXKxZC1W6Oj7G5KajvjHm=4Gr5FdQ@mail.gmail.com>
	<CAJwe+CCR_fA7i6jMCutEwF8e+c=mEWqV478t_AVB=D3N8Org=A@mail.gmail.com>
	<007701d15d0d$4dfba330$e9f2e990$@botecomm.com>
	<CAJ0sghRyA8YhuMiJGN=x9iV_sh1Q=kkoHzTkeUaAZ_XZaEKbWQ@mail.gmail.com>
Message-ID: <CACOYK=jTLW1=ZHEU4AMg+VAwZHv+zoTQxL73m65e1D4P0ToZyA@mail.gmail.com>

by the way, are there any known concerns in running FreeSWITCH inside
an LXC container?

LXC is really convenient when network separation is required.


From s.safarov at gmail.com  Tue Feb  2 12:56:25 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 2 Feb 2016 12:56:25 +0300
Subject: [Freeswitch-users] notes about amazon AWS?
In-Reply-To: <56B06C2D.5010505@ringme.ru>
References: <56B06C2D.5010505@ringme.ru>
Message-ID: <CAHtxdDf33EcjG5WVmeDb0W9NMo4FMF02Rhm+cMSfJw=HnMTCqQ@mail.gmail.com>

PV mode not supported for new AMI. Use only HWM


On Tue, Feb 2, 2016 at 11:43 AM, ????? <denis at ringme.ru> wrote:

> Hello, who can share AMI for freeswitch and maybe some notes about setup
> (best practices)? But need centos 6 x86-64, it's _requirement_ some
> other our software.
>
> PV or HWM mode?
> https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 - too
> few info about centos.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 stephen.thwaites at callstera.com  Tue Feb  2 13:05:14 2016
From: stephen.thwaites at callstera.com (Stephen Thwaites)
Date: Tue, 2 Feb 2016 11:05:14 +0100
Subject: [Freeswitch-users] BLF Subscriptions sometimes don't send an
	initial Notify
Message-ID: <CAB65EiveLwtrdLEbcdTUy3SumEV8N=D2ryTin4KVf_tc=o1Ppw@mail.gmail.com>

Hello,
I have setup presence and in most cases it is working as expected. i.e.
Subscription is sent to FS, FS returns Accepted then immediately FS sends
the notify to the phone, thereafter all Notifies for ringing, pickup and
hangup. Great.

However in some cases a subscribe to an extension does subscribe, FS sends
the accepted response but a Notify is not sent out at that point. However
if I call the extension the Notify works perfectly.

Any ideas of what could cause the initial Notify not to be sent after the
Acceptance 202?

Any help would be appreciated.

Regards,
Steve.

Some info below:
FS is configured as Multi-Tennant

<param name="presence-proto-lookup" value="false"/>
<param name="send-presence-on-register" value="true"/>
<param name="manage-presence" value="true"/>
<param name="presence-probe-on-register" value="true"/>
<param name="manage-shared-appearance" value="true"/>
<param name="dbname" value="freeswitch"/>
<param name="presence-hosts" value="_DISABLE_"/>   ** Multi-Tennant
<param name="presence-privacy" value="false"/>

SIP Trace for Subscription

FS Receives this from the phone:

SUBSCRIBE sip:203 at xxx.mydomain.com:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.24:25060;branch=z9hG4bK991003231;rport
From: <sip:201 at xxx.mydomain.com>;tag=946578510
To: <sip:203 at xxx.mydomain.com>;tag=nvqzSrr57RZg
Call-ID: 968288340-25060-7 at BJC.BGI.B.CE
CSeq: 20050 SUBSCRIBE
Contact: <sip:201 at 192.168.1.24:25060>
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GXP2140 1.0.5.29
Expires: 480
Supported: replaces, path, timer, eventlist
Event: dialog
Accept: application/dialog-info+xml,multipart/related,application/rlmi+xml
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE, MESSAGE
Content-Length: 0

FS Sends this back to the phone:

SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 192.168.1.24:25060
;branch=z9hG4bK991003231;rport=59364;received=x.x.x.x
From: <sip:201 at xxx.mydomain.com>;tag=946578510
To: <sip:203 at xxx.mydomain.com>;tag=nvqzSrr57RZg
Call-ID: 968288340-25060-7 at BJC.BGI.B.CE
CSeq: 20050 SUBSCRIBE
Contact: <sip:203 at y.y.y.y:5060>
Expires: 480
User-Agent: Callstera VOIP PBX v1.20
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer
Subscription-State: active;expires=480
Content-Length: 0
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From lists at telefaks.de  Tue Feb  2 14:40:37 2016
From: lists at telefaks.de (Peter Steinbach)
Date: Tue, 02 Feb 2016 12:40:37 +0100
Subject: [Freeswitch-users] FreeSWITCH in virtual environments
In-Reply-To: <CACOYK=jTLW1=ZHEU4AMg+VAwZHv+zoTQxL73m65e1D4P0ToZyA@mail.gmail.com>
References: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C9979B74@PHXEX2.vertical.com>
	<CAJ0sghSaQhKgD4X57DkEGP_MBxHgHi1vTLDh7E7tQXZtKRPz_A@mail.gmail.com>
	<88218521-FB01-4698-9DBF-625D99E8BC79@gmail.com>
	<CAHtxdDfjiAeSEZfnFSoAAK3syGMwBfFrV=Ox+np8AfdTteuqeg@mail.gmail.com>
	<58E2C5C4-87FF-4E01-8048-6C39B9362B02@gmail.com>
	<CAJwe+CBc64_0efV=-0RNr3kB_gtPPU6vTdbcxKWL7m+wGsmbVw@mail.gmail.com>
	<CAO_GGAoaFOg8cJ930DHbjyXKxZC1W6Oj7G5KajvjHm=4Gr5FdQ@mail.gmail.com>
	<CAJwe+CCR_fA7i6jMCutEwF8e+c=mEWqV478t_AVB=D3N8Org=A@mail.gmail.com>
	<007701d15d0d$4dfba330$e9f2e990$@botecomm.com>
	<CAJ0sghRyA8YhuMiJGN=x9iV_sh1Q=kkoHzTkeUaAZ_XZaEKbWQ@mail.gmail.com>
	<CACOYK=jTLW1=ZHEU4AMg+VAwZHv+zoTQxL73m65e1D4P0ToZyA@mail.gmail.com>
Message-ID: <56B095B5.8090804@telefaks.de>

Hello Stanislav,

we have just setup Freeswitch on Debian 8 inside LXC. We are examinating
to switch from OpenVZ zu LXC.

Freeswitch works so far productive, but with almost no load, so I have
no performance figures yet. We will do some load and stress tests during
this month to compare against OpenVZ.

We expect advantages compared to OpenVZ with a newer kernel and in
networking, let's see then where the drawbacks are.

Best regards
Peter

On 02/02/16 10:06, Stanislav Sinyagin wrote:
> by the way, are there any known concerns in running FreeSWITCH inside
> an LXC container?
>
> LXC is really convenient when network separation is required.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de



From deforceczt at gmail.com  Tue Feb  2 15:08:49 2016
From: deforceczt at gmail.com (Vladislav Ivanov)
Date: Tue, 2 Feb 2016 14:08:49 +0200
Subject: [Freeswitch-users] Does freeswitch forks his processes?
Message-ID: <CAMCT=3R8CLCX2YBbNghAbr6NouK08=MpXUB2T0puoZ-rhJz-uQ@mail.gmail.com>

Hey guys,

I have a question about freeswitch process/threading usage.
So far that I haven't noticed freeswitch to fork himself, I have only 1
freeswitch instance.
http://i.imgur.com/bdbYOwp.png

But then I found screenshot of htop with freeswitch and noticed that there
is multiple freeswitch processes being run:
http://i.imgur.com/VNpl55z.jpg

I'm having issues with "loading" the freeswitch after 50 cps in any cpu/ram
configuration.
Be it physical or virtual environment I cant pass the 50 cps mark.
I have strange issue with CPU usage on same CPS:

http://i.imgur.com/8BdQWVL.png
http://i.imgur.com/mWRnoGr.png

I timeload test freeswitch with 50cps for 5+ hours, and seems like there is
some kind of leak somewhere.
I have tested configuration on:
Debian 8
2 core/8 gb ram
4 core/8 gb ram (graphs are from here)
8 core/32 gb ram

and in all the tests I were not able to send more than 50 cps without CPU
dropping to 0 with all system starting to respond really laggy.

Test is:
sipp -> freeswitch -> sipp

Just 1 dialpeer with bridge action. No gateways. Just simple dialplan and 1
profile...
Any advice?

Thank you all
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From gregor at infomedia.si  Tue Feb  2 11:50:14 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Tue, 2 Feb 2016 09:50:14 +0100
Subject: [Freeswitch-users] TCP registrations
In-Reply-To: <001b01d15d51$455900d0$d00b0270$@botecomm.com>
References: <loom.20160201T233412-829@post.gmane.org>
	<001b01d15d51$455900d0$d00b0270$@botecomm.com>
Message-ID: <CAC7ttFGGnvRDPHkCSDjwt2Ob0HXvBZv+_FXHCwXj8GFHHCuWHw@mail.gmail.com>

Yes, I also think so, but cannot find explicitly documented. So please, if
anyone know exactly which command is, please help.

2016-02-02 1:32 GMT+01:00 Bote Man <bote_radio at botecomm.com>:

> FreeSWITCH uses UDP by default for SIP signaling. You can change this in
> the
> SIP_profile I believe.
>
>
> ---
> Bote
>
> FreeSWITCH Docs Janitor
> http://freeswitch.org/confluence
>
>
>
>
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> > users-bounces at lists.freeswitch.org] On Behalf Of Gregor
> > Sent: Monday, 01 February, 2016 17:37
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: [Freeswitch-users] TCP registrations
> >
> > I think I am missing something.
> >
> > I would like to configure freeswitch that listens on TCP port for client
> > registrations (internal profile). As I read, freeswitch should do this by
> > default. But freeswitch responses only on UDP protocol. Is there a conf
> > setting for specify also tcp for registrations.
> >
> >
> > __________________________________________________________
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 roman at dissauer.net  Tue Feb  2 14:58:22 2016
From: roman at dissauer.net (Roman Dissauer)
Date: Tue, 2 Feb 2016 12:58:22 +0100
Subject: [Freeswitch-users] BLF for Gateway
Message-ID: <BD20678C-416F-413D-AC6E-077C888C3745@dissauer.net>

Hi All,

is there a way to get gateway usage in freeswitch on my phones blf?
I have multiple gateways registered and want to see which one is taken for a particular outbound call.

Best Regards,
Roman


From bote_radio at botecomm.com  Tue Feb  2 15:49:17 2016
From: bote_radio at botecomm.com (Bote Man)
Date: Tue, 2 Feb 2016 07:49:17 -0500
Subject: [Freeswitch-users] FreeSWITCH in virtual environments
In-Reply-To: <56B095B5.8090804@telefaks.de>
References: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C9979B74@PHXEX2.vertical.com>	<CAJ0sghSaQhKgD4X57DkEGP_MBxHgHi1vTLDh7E7tQXZtKRPz_A@mail.gmail.com>	<88218521-FB01-4698-9DBF-625D99E8BC79@gmail.com>	<CAHtxdDfjiAeSEZfnFSoAAK3syGMwBfFrV=Ox+np8AfdTteuqeg@mail.gmail.com>	<58E2C5C4-87FF-4E01-8048-6C39B9362B02@gmail.com>	<CAJwe+CBc64_0efV=-0RNr3kB_gtPPU6vTdbcxKWL7m+wGsmbVw@mail.gmail.com>	<CAO_GGAoaFOg8cJ930DHbjyXKxZC1W6Oj7G5KajvjHm=4Gr5FdQ@mail.gmail.com>	<CAJwe+CCR_fA7i6jMCutEwF8e+c=mEWqV478t_AVB=D3N8Org=A@mail.gmail.com>	<007701d15d0d$4dfba330$e9f2e990$@botecomm.com>	<CAJ0sghRyA8YhuMiJGN=x9iV_sh1Q=kkoHzTkeUaAZ_XZaEKbWQ@mail.gmail.com>	<CACOYK=jTLW1=ZHEU4AMg+VAwZHv+zoTQxL73m65e1D4P0ToZyA@mail.gmail.com>
	<56B095B5.8090804@telefaks.de>
Message-ID: <005601d15db8$219762b0$64c62810$@botecomm.com>

Peter, I / we would be grateful if you would kindly update us with your
experiences on this page
https://freeswitch.org/confluence/display/FREESWITCH/Virtualization+Experien
ces
as well as the mailing list.

Because each installation is different it is helpful to compile
configurations that work and don't work where others can benefit from these
experiences.

Vielen Dank!

---
Bote
FreeSWITCH Docs Janitor
http://freeswitch.org/confluence


> -----Original Message-----
> From: Peter Steinbach
> Sent: Tuesday, 02 February, 2016 06:41
> Subject: Re: [Freeswitch-users] FreeSWITCH in virtual environments
> 
> Hello Stanislav,
> 
> we have just setup Freeswitch on Debian 8 inside LXC. We are examinating
> to switch from OpenVZ zu LXC.
> 
> Freeswitch works so far productive, but with almost no load, so I have
> no performance figures yet. We will do some load and stress tests during
> this month to compare against OpenVZ.
> 
> We expect advantages compared to OpenVZ with a newer kernel and in
> networking, let's see then where the drawbacks are.
> 
> Best regards
> Peter
> 
> On 02/02/16 10:06, Stanislav Sinyagin wrote:
> > by the way, are there any known concerns in running FreeSWITCH inside
> > an LXC container?
> >
> > LXC is really convenient when network separation is required.
> >
> >
> __________________________________________________________
> _______________



From lists at plustel.dk  Tue Feb  2 15:21:43 2016
From: lists at plustel.dk (Tom Braarup Cuykens)
Date: Tue, 2 Feb 2016 13:21:43 +0100
Subject: [Freeswitch-users] FreeSWITCH in virtual environments
In-Reply-To: <56B095B5.8090804@telefaks.de>
References: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C9979B74@PHXEX2.vertical.com>
	<CAJ0sghSaQhKgD4X57DkEGP_MBxHgHi1vTLDh7E7tQXZtKRPz_A@mail.gmail.com>
	<88218521-FB01-4698-9DBF-625D99E8BC79@gmail.com>
	<CAHtxdDfjiAeSEZfnFSoAAK3syGMwBfFrV=Ox+np8AfdTteuqeg@mail.gmail.com>
	<58E2C5C4-87FF-4E01-8048-6C39B9362B02@gmail.com>
	<CAJwe+CBc64_0efV=-0RNr3kB_gtPPU6vTdbcxKWL7m+wGsmbVw@mail.gmail.com>
	<CAO_GGAoaFOg8cJ930DHbjyXKxZC1W6Oj7G5KajvjHm=4Gr5FdQ@mail.gmail.com>
	<CAJwe+CCR_fA7i6jMCutEwF8e+c=mEWqV478t_AVB=D3N8Org=A@mail.gmail.com>
	<007701d15d0d$4dfba330$e9f2e990$@botecomm.com>
	<CAJ0sghRyA8YhuMiJGN=x9iV_sh1Q=kkoHzTkeUaAZ_XZaEKbWQ@mail.gmail.com>
	<CACOYK=jTLW1=ZHEU4AMg+VAwZHv+zoTQxL73m65e1D4P0ToZyA@mail.gmail.com>
	<56B095B5.8090804@telefaks.de>
Message-ID: <56B09F57.2010801@plustel.dk>

Hello Peter,


This is very interesting. Would love to have feedback and recipe to see 
how it worked out.

Kind Regards,

Tom Braarup Cuykens



On 02/02/2016 12:40 PM, Peter Steinbach wrote:
> Hello Stanislav,
>
> we have just setup Freeswitch on Debian 8 inside LXC. We are examinating
> to switch from OpenVZ zu LXC.
>
> Freeswitch works so far productive, but with almost no load, so I have
> no performance figures yet. We will do some load and stress tests during
> this month to compare against OpenVZ.
>
> We expect advantages compared to OpenVZ with a newer kernel and in
> networking, let's see then where the drawbacks are.
>
> Best regards
> Peter
>
> On 02/02/16 10:06, Stanislav Sinyagin wrote:
>> by the way, are there any known concerns in running FreeSWITCH inside
>> an LXC container?
>>
>> LXC is really convenient when network separation is required.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 royj at yandex.ru  Tue Feb  2 15:54:43 2016
From: royj at yandex.ru (royj at yandex.ru)
Date: Tue, 02 Feb 2016 15:54:43 +0300
Subject: [Freeswitch-users] async, sendmsg, execute-app-name
Message-ID: <4661454417683@web12o.yandex.ru>

Hi, all

I am working on nodejs outbound socket application and faced that if send to FreeSWITCH message like:

sendmsg
execute-app-name: playback
execute-app-arg: silence_stream://2000
call-command: execute

when there is already no corresponding channel (a caller hung up by itself and from FreesSWITCH received CHANNEL_HANGUP_COMPLETE, text/disconnect-notice;linger)

FreesSWITCH answers:

Content-Type: command/reply
Reply-Text: +OK

Logic of library and application such, that we know that an application is finished, when received CHANNEL_EXECUTE_COMPLETE.
There are cases when a caller hung up and exactly in this moment application not yet knows about it and sends message 'call-command: execute' and then there is javascript Promise in pending state, seems like forever.
Is there any chance to understand in the answer that the channel does not exist?


From bote_radio at botecomm.com  Tue Feb  2 15:58:46 2016
From: bote_radio at botecomm.com (Bote Man)
Date: Tue, 2 Feb 2016 07:58:46 -0500
Subject: [Freeswitch-users] Does freeswitch forks his processes?
In-Reply-To: <CAMCT=3R8CLCX2YBbNghAbr6NouK08=MpXUB2T0puoZ-rhJz-uQ@mail.gmail.com>
References: <CAMCT=3R8CLCX2YBbNghAbr6NouK08=MpXUB2T0puoZ-rhJz-uQ@mail.gmail.com>
Message-ID: <005d01d15db9$744a3bd0$5cdeb370$@botecomm.com>

Is FreeSWITCH starting with root permissions? It needs this in order to use the FIFO scheduler and access realtime threads. If not started as root, this would explain your CPS limitations. There are also limits that can be set in the config files.

 

After it starts it drops privileges to those specified on the command line with ?u and ?g switches.

 

FreeSWITCH uses multi-threading. I do not know about htop, but maybe it is showing the multiple threads?

 

top ?H shows each thread.

 

---

Bote

FreeSWITCH Docs Janitor

 <http://freeswitch.org/confluence> http://freeswitch.org/confluence

 

 

 

 

 

From: Vladislav Ivanov
Sent: Tuesday, 02 February, 2016 07:09
Subject: [Freeswitch-users] Does freeswitch forks his processes?

 

Hey guys,

I have a question about freeswitch process/threading usage.
So far that I haven't noticed freeswitch to fork himself, I have only 1 freeswitch instance.
http://i.imgur.com/bdbYOwp.png

But then I found screenshot of htop with freeswitch and noticed that there is multiple freeswitch processes being run:
http://i.imgur.com/VNpl55z.jpg

I'm having issues with "loading" the freeswitch after 50 cps in any cpu/ram configuration.
Be it physical or virtual environment I cant pass the 50 cps mark.
I have strange issue with CPU usage on same CPS:

http://i.imgur.com/8BdQWVL.png
http://i.imgur.com/mWRnoGr.png

I timeload test freeswitch with 50cps for 5+ hours, and seems like there is some kind of leak somewhere.
I have tested configuration on:
Debian 8
2 core/8 gb ram
4 core/8 gb ram (graphs are from here)
8 core/32 gb ram

and in all the tests I were not able to send more than 50 cps without CPU dropping to 0 with all system starting to respond really laggy.

Test is:
sipp -> freeswitch -> sipp

Just 1 dialpeer with bridge action. No gateways. Just simple dialplan and 1 profile...
Any advice?

Thank you all





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From giacomo.vacca at gmail.com  Tue Feb  2 15:59:19 2016
From: giacomo.vacca at gmail.com (Giacomo Vacca)
Date: Tue, 2 Feb 2016 13:59:19 +0100
Subject: [Freeswitch-users] Does freeswitch forks his processes?
In-Reply-To: <CAMCT=3R8CLCX2YBbNghAbr6NouK08=MpXUB2T0puoZ-rhJz-uQ@mail.gmail.com>
References: <CAMCT=3R8CLCX2YBbNghAbr6NouK08=MpXUB2T0puoZ-rhJz-uQ@mail.gmail.com>
Message-ID: <CAO6UKa2GgrWogXTTU6KY+hXSFQfVhj8t=XYF5aTp2+FJx-d1jA@mail.gmail.com>

FreeSWITCH is a single process, multi-threading application.
htop is showing you threads activity.

On 2 February 2016 at 13:08, Vladislav Ivanov <deforceczt at gmail.com> wrote:

> Hey guys,
>
> I have a question about freeswitch process/threading usage.
> So far that I haven't noticed freeswitch to fork himself, I have only 1
> freeswitch instance.
> http://i.imgur.com/bdbYOwp.png
>
> But then I found screenshot of htop with freeswitch and noticed that there
> is multiple freeswitch processes being run:
> http://i.imgur.com/VNpl55z.jpg
>
> I'm having issues with "loading" the freeswitch after 50 cps in any
> cpu/ram configuration.
> Be it physical or virtual environment I cant pass the 50 cps mark.
> I have strange issue with CPU usage on same CPS:
>
> http://i.imgur.com/8BdQWVL.png
> http://i.imgur.com/mWRnoGr.png
>
> I timeload test freeswitch with 50cps for 5+ hours, and seems like there
> is some kind of leak somewhere.
> I have tested configuration on:
> Debian 8
> 2 core/8 gb ram
> 4 core/8 gb ram (graphs are from here)
> 8 core/32 gb ram
>
> and in all the tests I were not able to send more than 50 cps without CPU
> dropping to 0 with all system starting to respond really laggy.
>
> Test is:
> sipp -> freeswitch -> sipp
>
> Just 1 dialpeer with bridge action. No gateways. Just simple dialplan and
> 1 profile...
> Any advice?
>
> Thank you all
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb  2 16:01:23 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 2 Feb 2016 16:01:23 +0300
Subject: [Freeswitch-users] TCP registrations
In-Reply-To: <CAC7ttFGGnvRDPHkCSDjwt2Ob0HXvBZv+_FXHCwXj8GFHHCuWHw@mail.gmail.com>
References: <loom.20160201T233412-829@post.gmane.org>
	<001b01d15d51$455900d0$d00b0270$@botecomm.com>
	<CAC7ttFGGnvRDPHkCSDjwt2Ob0HXvBZv+_FXHCwXj8GFHHCuWHw@mail.gmail.com>
Message-ID: <CAHtxdDeBNmV0KVxw-nL-BsX-c-q1UdRmxpFXoPuo6sjo9f0L+A@mail.gmail.com>

FS is defailt support UDP, TCP tansport. If you enable TLS in vars.xml also
TLS transport is will be enabled.
To check what is type of socket is open on server please use
netstat -an --inet | grep -P "5060|5061|5080"

Example
[admin at node1.sbc ~]$ netstat -an --inet | grep -P "5060|5061|5080"
tcp        0      0 217.12.247.214:5060     0.0.0.0:*               LISTEN

tcp        0      0 10.21.7.30:5060         0.0.0.0:*               LISTEN

tcp        0      0 217.12.247.214:5061     0.0.0.0:*               LISTEN

tcp        0      0 217.12.247.214:5080     0.0.0.0:*               LISTEN

udp        0      0 217.12.247.214:5060     0.0.0.0:*

udp        0      0 10.21.7.30:5060         0.0.0.0:*

udp        0      0 217.12.247.214:5080     0.0.0.0:*

On Tue, Feb 2, 2016 at 11:50 AM, Gregor Nanger <gregor at infomedia.si> wrote:

> Yes, I also think so, but cannot find explicitly documented. So please, if
> anyone know exactly which command is, please help.
>
> 2016-02-02 1:32 GMT+01:00 Bote Man <bote_radio at botecomm.com>:
>
>> FreeSWITCH uses UDP by default for SIP signaling. You can change this in
>> the
>> SIP_profile I believe.
>>
>>
>> ---
>> Bote
>>
>> FreeSWITCH Docs Janitor
>> http://freeswitch.org/confluence
>>
>>
>>
>>
>> > -----Original Message-----
>> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
>> > users-bounces at lists.freeswitch.org] On Behalf Of Gregor
>> > Sent: Monday, 01 February, 2016 17:37
>> > To: freeswitch-users at lists.freeswitch.org
>> > Subject: [Freeswitch-users] TCP registrations
>> >
>> > I think I am missing something.
>> >
>> > I would like to configure freeswitch that listens on TCP port for client
>> > registrations (internal profile). As I read, freeswitch should do this
>> by
>> > default. But freeswitch responses only on UDP protocol. Is there a conf
>> > setting for specify also tcp for registrations.
>> >
>> >
>> > __________________________________________________________
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 asilva at wirelessmundi.com  Tue Feb  2 16:19:14 2016
From: asilva at wirelessmundi.com (Antonio Silva)
Date: Tue, 2 Feb 2016 14:19:14 +0100
Subject: [Freeswitch-users] TCP registrations
In-Reply-To: <CAHtxdDeBNmV0KVxw-nL-BsX-c-q1UdRmxpFXoPuo6sjo9f0L+A@mail.gmail.com>
References: <loom.20160201T233412-829@post.gmane.org>
	<001b01d15d51$455900d0$d00b0270$@botecomm.com>
	<CAC7ttFGGnvRDPHkCSDjwt2Ob0HXvBZv+_FXHCwXj8GFHHCuWHw@mail.gmail.com>
	<CAHtxdDeBNmV0KVxw-nL-BsX-c-q1UdRmxpFXoPuo6sjo9f0L+A@mail.gmail.com>
Message-ID: <56B0ACD2.9070800@wirelessmundi.com>

The parameter is "bind-params"

https://freeswitch.org/confluence/display/FREESWITCH/Sofia+Configuration+Files

by default fs should bind to tcp and udp but if you want only tcp just 
set for the profile:

<param name="bind-params" value="transport=tcp"/>



On 02/02/2016 02:01 PM, Sergey Safarov wrote:
> FS is defailt support UDP, TCP tansport. If you enable TLS in vars.xml 
> also TLS transport is will be enabled.
> To check what is type of socket is open on server please use
> netstat -an --inet | grep -P "5060|5061|5080"
>
> Example
> [admin at node1.sbc ~]$ netstat -an --inet | grep -P "5060|5061|5080"
> tcp        0      0 217.12.247.214:5060 <http://217.12.247.214:5060>   
>   0.0.0.0:*               LISTEN
> tcp        0      0 10.21.7.30:5060 <http://10.21.7.30:5060> 0.0.0.0:* 
>               LISTEN
> tcp        0      0 217.12.247.214:5061 <http://217.12.247.214:5061>   
>   0.0.0.0:*               LISTEN
> tcp        0      0 217.12.247.214:5080 <http://217.12.247.214:5080>   
>   0.0.0.0:*               LISTEN
> udp        0      0 217.12.247.214:5060 <http://217.12.247.214:5060>   
>   0.0.0.0:*
> udp        0      0 10.21.7.30:5060 <http://10.21.7.30:5060> 0.0.0.0:*
> udp        0      0 217.12.247.214:5080 <http://217.12.247.214:5080>   
>   0.0.0.0:*
>
> On Tue, Feb 2, 2016 at 11:50 AM, Gregor Nanger <gregor at infomedia.si 
> <mailto:gregor at infomedia.si>> wrote:
>
>     Yes, I also think so, but cannot find explicitly documented. So
>     please, if anyone know exactly which command is, please help.
>
>     2016-02-02 1:32 GMT+01:00 Bote Man <bote_radio at botecomm.com
>     <mailto:bote_radio at botecomm.com>>:
>
>         FreeSWITCH uses UDP by default for SIP signaling. You can
>         change this in the
>         SIP_profile I believe.
>
>
>         ---
>         Bote
>
>         FreeSWITCH Docs Janitor
>         http://freeswitch.org/confluence
>
>
>
>
>         > -----Original Message-----
>         > From: freeswitch-users-bounces at lists.freeswitch.org
>         <mailto:freeswitch-users-bounces at lists.freeswitch.org>
>         [mailto:freeswitch- <mailto:freeswitch->
>         > users-bounces at lists.freeswitch.org
>         <mailto:users-bounces at lists.freeswitch.org>] On Behalf Of Gregor
>         > Sent: Monday, 01 February, 2016 17:37
>         > To: freeswitch-users at lists.freeswitch.org
>         <mailto:freeswitch-users at lists.freeswitch.org>
>         > Subject: [Freeswitch-users] TCP registrations
>         >
>         > I think I am missing something.
>         >
>         > I would like to configure freeswitch that listens on TCP
>         port for client
>         > registrations (internal profile). As I read, freeswitch
>         should do this by
>         > default. But freeswitch responses only on UDP protocol. Is
>         there a conf
>         > setting for specify also tcp for registrations.
>         >
>         >
>         > __________________________________________________________
>
>
>         _________________________________________________________________________
>         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
>
>
>
>
>     -- 
>     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
>
>     Official 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

-- 

Saludos / Regards / Cumprimentos,
Ant?nio silva

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From gb at cm.nl  Tue Feb  2 16:37:28 2016
From: gb at cm.nl (Grant Bagdasarian)
Date: Tue, 2 Feb 2016 13:37:28 +0000
Subject: [Freeswitch-users] FreeSWITCH in virtual environments
In-Reply-To: <CAJwe+CBc64_0efV=-0RNr3kB_gtPPU6vTdbcxKWL7m+wGsmbVw@mail.gmail.com>
References: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C9979B74@PHXEX2.vertical.com>
	<CAJ0sghSaQhKgD4X57DkEGP_MBxHgHi1vTLDh7E7tQXZtKRPz_A@mail.gmail.com>
	<88218521-FB01-4698-9DBF-625D99E8BC79@gmail.com>
	<CAHtxdDfjiAeSEZfnFSoAAK3syGMwBfFrV=Ox+np8AfdTteuqeg@mail.gmail.com>
	<58E2C5C4-87FF-4E01-8048-6C39B9362B02@gmail.com>
	<CAJwe+CBc64_0efV=-0RNr3kB_gtPPU6vTdbcxKWL7m+wGsmbVw@mail.gmail.com>
Message-ID: <1171818c8a824ad2ad7f37a0688f39a2@CM-EX-V01.cm.local>

We never tested with 300 CPS, but the host itself has 8 GB of memory.
Our VM?s usually have 2-4 CPU?s and 1-2 GB of memory, but memory is hardly used. It?s CPU you need.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of E. Schmidbauer
Sent: Monday, February 1, 2016 4:17 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] FreeSWITCH in virtual environments

Can anyone share the VM memory/cpu specs used in these cases?
We want to run around 300 CPS on FS (running on vmware).
very little transcoding (if any), audio only
How much memory/cpu should be provisioned?
I see Grant mentioned "single VM host (6 Cores, 12 Threads)" but how much memory?
Thanks,
E

On Fri, Jan 29, 2016 at 10:00 PM, servtelar <servtelar at gmail.com<mailto:servtelar at gmail.com>> wrote:
Thanks a lot guys for sharing this info. It?s really helpful.


On Jan 28, 2016, at 6:18 PM, Sergey Safarov <s.safarov at gmail.com<mailto:s.safarov at gmail.com>> wrote:

We have to core ESXi vm with 140 session (70 calls) with have 70 CPU load.

Sergey

On Fri, Jan 29, 2016 at 2:01 AM, servtelar <servtelar at gmail.com<mailto:servtelar at gmail.com>> wrote:
Hi Chad

How many legs you are handling with 20 cores on a conference?

Regards

Gustavo

On Jan 28, 2016, at 7:55 PM, Chad Phillips <chad at apartmentlines.com<mailto:chad at apartmentlines.com>> wrote:

I've had very good luck running the newer video branch code on ProfitBricks: https://www.profitbricks.com/

As far as I understand, the CPU cycles are guaranteed on their platform. I've had to put as many as 20 cores on a server to handle some of our busier video conference calls, but with that it runs quite smoothly.

On Thu, Jan 28, 2016 at 2:15 PM, Dan Edwards <DEdwards at vertical.com<mailto:DEdwards at vertical.com>> wrote:
I am reviewing the Confluence Virtualization page and had some questions, in particular about VMWare. My company distributes some of its software as a VMWare image file and we were looking to distribute a new product using FS in the same manner. The products operate at a customer premise, on their VMWare infrastructure, not in a cloud environment. Since our customers already have VMWare, switching to a different VM infrastructure is going to hurt, so I am looking for options/alternatives.

First, does anybody know if the virtual timing issues with VMWare have improved since this page was last updated in 2014? Is VMWare still not good enough? Is it possible to throw CPU & memory at this and make VMWare good enough, or is the virtual timing just not workable?

On the virtualization page, there was a comment from 2010 that you might be happy with a High CPU Medium instance on AWS EC2. Certainly workload is a factor here, but I am trying to get my head around how big a machine to perform how small a workload. Is there a place where people talk about their experiences?

Are there other VM platforms that might be acceptable?

Any help or comment is appreciated.

Thank you,
Dan


_________________________________________________________________________
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<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 krice at freeswitch.org  Tue Feb  2 17:53:10 2016
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 2 Feb 2016 08:53:10 -0600
Subject: [Freeswitch-users] Does freeswitch forks his processes?
In-Reply-To: <CAMCT=3R8CLCX2YBbNghAbr6NouK08=MpXUB2T0puoZ-rhJz-uQ@mail.gmail.com>
References: <CAMCT=3R8CLCX2YBbNghAbr6NouK08=MpXUB2T0puoZ-rhJz-uQ@mail.gmail.com>
Message-ID: <555b01d15dc9$70137cb0$503a7610$@freeswitch.org>

FreeSWITCH uses threading?  in modern linux kernels threading and forking are very similar? if you look in top you?ll only see 1 FS process that?s because top by default rolls up all the threads? htop on the other hand by default shows you the individual threads.

 

In FreeSWITCH there are several threads running on just a base idle FreeSWITCH process. Each addition call leg is atleast 1 more thread. Depending on which applications are actually active on a call, there could be more then 1 thread per call leg.

 

 

On the load testing and load handling capabilities this is something the FS team typically does not support on the open source side. I would recommend contacting consulting at freeswitch.org <mailto:consulting at freeswitch.org>  to get the pros involved. There are a lot of factors to take into consideration and just 1 param can make orders of magnitude difference from test to test

 

K

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Vladislav Ivanov
Sent: Tuesday, February 2, 2016 6:09 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Does freeswitch forks his processes?

 

Hey guys,

I have a question about freeswitch process/threading usage.
So far that I haven't noticed freeswitch to fork himself, I have only 1 freeswitch instance.
http://i.imgur.com/bdbYOwp.png

But then I found screenshot of htop with freeswitch and noticed that there is multiple freeswitch processes being run:
http://i.imgur.com/VNpl55z.jpg

I'm having issues with "loading" the freeswitch after 50 cps in any cpu/ram configuration.
Be it physical or virtual environment I cant pass the 50 cps mark.
I have strange issue with CPU usage on same CPS:

http://i.imgur.com/8BdQWVL.png
http://i.imgur.com/mWRnoGr.png

I timeload test freeswitch with 50cps for 5+ hours, and seems like there is some kind of leak somewhere.
I have tested configuration on:
Debian 8
2 core/8 gb ram
4 core/8 gb ram (graphs are from here)
8 core/32 gb ram

and in all the tests I were not able to send more than 50 cps without CPU dropping to 0 with all system starting to respond really laggy.

Test is:
sipp -> freeswitch -> sipp

Just 1 dialpeer with bridge action. No gateways. Just simple dialplan and 1 profile...
Any advice?

Thank you all





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From lists at telefaks.de  Tue Feb  2 18:10:54 2016
From: lists at telefaks.de (Peter Steinbach)
Date: Tue, 02 Feb 2016 16:10:54 +0100
Subject: [Freeswitch-users] Does freeswitch forks his processes?
In-Reply-To: <005d01d15db9$744a3bd0$5cdeb370$@botecomm.com>
References: <CAMCT=3R8CLCX2YBbNghAbr6NouK08=MpXUB2T0puoZ-rhJz-uQ@mail.gmail.com>
	<005d01d15db9$744a3bd0$5cdeb370$@botecomm.com>
Message-ID: <56B0C6FE.9070507@telefaks.de>

I've just stumpled over this:
>Is FreeSWITCH starting with root permissions? It needs this in order to
use the FIFO scheduler and access realtime threads. If not started as
root, this would explain your CPS limitations.

We like to run Freeswitch as a non privileged user, due to security
concerns. So there are drawbacks here compared to running FS as root?
Can we somehow quantify the differences?

Best regards
Peter


On 02/02/16 13:58, Bote Man wrote:
>
> Is FreeSWITCH starting with root permissions? It needs this in order
> to use the FIFO scheduler and access realtime threads. If not started
> as root, this would explain your CPS limitations. There are also
> limits that can be set in the config files.
>
>  
>
> After it starts it drops privileges to those specified on the command
> line with --u and --g switches.
>
>  
>
> FreeSWITCH uses multi-threading. I do not know about htop, but maybe
> it is showing the multiple threads?
>
>  
>
> top --H shows each thread.
>
>  
>
> ---
>
> Bote
>
> FreeSWITCH Docs Janitor
>
> http://freeswitch.org/confluence
>
>  
>
>  
>
>  
>
>  
>
>  
>
> *From:*Vladislav Ivanov
> *Sent:* Tuesday, 02 February, 2016 07:09
> *Subject:* [Freeswitch-users] Does freeswitch forks his processes?
>
>  
>
> Hey guys,
>
> I have a question about freeswitch process/threading usage.
> So far that I haven't noticed freeswitch to fork himself, I have only
> 1 freeswitch instance.
> http://i.imgur.com/bdbYOwp.png
>
> But then I found screenshot of htop with freeswitch and noticed that
> there is multiple freeswitch processes being run:
> http://i.imgur.com/VNpl55z.jpg
>
> I'm having issues with "loading" the freeswitch after 50 cps in any
> cpu/ram configuration.
> Be it physical or virtual environment I cant pass the 50 cps mark.
> I have strange issue with CPU usage on same CPS:
>
> http://i.imgur.com/8BdQWVL.png
> http://i.imgur.com/mWRnoGr.png
>
> I timeload test freeswitch with 50cps for 5+ hours, and seems like
> there is some kind of leak somewhere.
> I have tested configuration on:
> Debian 8
> 2 core/8 gb ram
> 4 core/8 gb ram (graphs are from here)
> 8 core/32 gb ram
>
> and in all the tests I were not able to send more than 50 cps without
> CPU dropping to 0 with all system starting to respond really laggy.
>
> Test is:
> sipp -> freeswitch -> sipp
>
> Just 1 dialpeer with bridge action. No gateways. Just simple dialplan
> and 1 profile...
> Any advice?
>
> Thank you all
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de

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From nneul at mst.edu  Tue Feb  2 18:22:31 2016
From: nneul at mst.edu (Nathan Neulinger)
Date: Tue, 2 Feb 2016 09:22:31 -0600
Subject: [Freeswitch-users] Does freeswitch forks his processes?
In-Reply-To: <56B0C6FE.9070507@telefaks.de>
References: <CAMCT=3R8CLCX2YBbNghAbr6NouK08=MpXUB2T0puoZ-rhJz-uQ@mail.gmail.com>
	<005d01d15db9$744a3bd0$5cdeb370$@botecomm.com>
	<56B0C6FE.9070507@telefaks.de>
Message-ID: <56B0C9B7.8000106@mst.edu>

"Run as" != "Start as"

If you insist on not starting FS as root to let it change user, like most other daemons/services, you'll have to jump 
through a bunch of extra steps using file system capabilities to give it the ability to set scheduler parameters/etc 
that are restricted to root normally.

-- Nathan

On 02/02/2016 09:10 AM, Peter Steinbach wrote:
> I've just stumpled over this:
>  >Is FreeSWITCH starting with root permissions? It needs this in order to use the FIFO scheduler and access realtime
> threads. If not started as root, this would explain your CPS limitations.
>
> We like to run Freeswitch as a non privileged user, due to security concerns. So there are drawbacks here compared to
> running FS as root? Can we somehow quantify the differences?
>
> Best regards
> Peter
>
>
> On 02/02/16 13:58, Bote Man wrote:
>>
>> Is FreeSWITCH starting with root permissions? It needs this in order to use the FIFO scheduler and access realtime
>> threads. If not started as root, this would explain your CPS limitations. There are also limits that can be set in the
>> config files.
>>
>> After it starts it drops privileges to those specified on the command line with ?u and ?g switches.
>>
>> FreeSWITCH uses multi-threading. I do not know about htop, but maybe it is showing the multiple threads?
>>
>> top ?H shows each thread.
>>
>> ---
>>
>> Bote
>>
>> FreeSWITCH Docs Janitor
>>
>> http://freeswitch.org/confluence
>>
>> *From:*Vladislav Ivanov
>> *Sent:* Tuesday, 02 February, 2016 07:09
>> *Subject:* [Freeswitch-users] Does freeswitch forks his processes?
>>
>> Hey guys,
>>
>> I have a question about freeswitch process/threading usage.
>> So far that I haven't noticed freeswitch to fork himself, I have only 1 freeswitch instance.
>> http://i.imgur.com/bdbYOwp.png
>>
>> But then I found screenshot of htop with freeswitch and noticed that there is multiple freeswitch processes being run:
>> http://i.imgur.com/VNpl55z.jpg
>>
>> I'm having issues with "loading" the freeswitch after 50 cps in any cpu/ram configuration.
>> Be it physical or virtual environment I cant pass the 50 cps mark.
>> I have strange issue with CPU usage on same CPS:
>>
>> http://i.imgur.com/8BdQWVL.png
>> http://i.imgur.com/mWRnoGr.png
>>
>> I timeload test freeswitch with 50cps for 5+ hours, and seems like there is some kind of leak somewhere.
>> I have tested configuration on:
>> Debian 8
>> 2 core/8 gb ram
>> 4 core/8 gb ram (graphs are from here)
>> 8 core/32 gb ram
>>
>> and in all the tests I were not able to send more than 50 cps without CPU dropping to 0 with all system starting to
>> respond really laggy.
>>
>> Test is:
>> sipp -> freeswitch -> sipp
>>
>> Just 1 dialpeer with bridge action. No gateways. Just simple dialplan and 1 profile...
>> Any advice?
>>
>> Thank you all
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> --
> With kind regards
> Peter Steinbach
>
> Telefaks Services GmbH
> mailto:lists  (att) telefaks.de
> Internet:www.telefaks.de
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 brian at freeswitch.org  Tue Feb  2 20:07:59 2016
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Feb 2016 11:07:59 -0600
Subject: [Freeswitch-users] TCP registrations
In-Reply-To: <56B0ACD2.9070800@wirelessmundi.com>
References: <loom.20160201T233412-829@post.gmane.org>
	<001b01d15d51$455900d0$d00b0270$@botecomm.com>
	<CAC7ttFGGnvRDPHkCSDjwt2Ob0HXvBZv+_FXHCwXj8GFHHCuWHw@mail.gmail.com>
	<CAHtxdDeBNmV0KVxw-nL-BsX-c-q1UdRmxpFXoPuo6sjo9f0L+A@mail.gmail.com>
	<56B0ACD2.9070800@wirelessmundi.com>
Message-ID: <CAEJMVkB9vt4xGPo3Eeqke2C_MRJY9nJzqoURetUKg3ArP+sTLQ@mail.gmail.com>

I'm going to guess your device probably fails to send the transport=tcp on
the contact there for it probably registers over TCP but we contact it back
over UDP?  Can you confirm?

On Tue, Feb 2, 2016 at 7:19 AM, Antonio Silva <asilva at wirelessmundi.com>
wrote:

> The parameter is "bind-params"
>
>
> https://freeswitch.org/confluence/display/FREESWITCH/Sofia+Configuration+Files
>
> by default fs should bind to tcp and udp but if you want only tcp just set
> for the profile:
>
> <param name="bind-params" value="transport=tcp"/>
>
>
>
>
> On 02/02/2016 02:01 PM, Sergey Safarov wrote:
>
> FS is defailt support UDP, TCP tansport. If you enable TLS in vars.xml
> also TLS transport is will be enabled.
> To check what is type of socket is open on server please use
> netstat -an --inet | grep -P "5060|5061|5080"
>
> Example
> [admin at node1.sbc ~]$ netstat -an --inet | grep -P "5060|5061|5080"
> tcp        0      0 217.12.247.214:5060     0.0.0.0:*
> LISTEN
> tcp        0      0 10.21.7.30:5060         0.0.0.0:*
> LISTEN
> tcp        0      0 217.12.247.214:5061     0.0.0.0:*
> LISTEN
> tcp        0      0 217.12.247.214:5080     0.0.0.0:*
> LISTEN
> udp        0      0 217.12.247.214:5060     0.0.0.0:*
>
> udp        0      0 10.21.7.30:5060         0.0.0.0:*
>
> udp        0      0 217.12.247.214:5080     0.0.0.0:*
>
> On Tue, Feb 2, 2016 at 11:50 AM, Gregor Nanger <gregor at infomedia.si>
> wrote:
>
>> Yes, I also think so, but cannot find explicitly documented. So please,
>> if anyone know exactly which command is, please help.
>>
>> 2016-02-02 1:32 GMT+01:00 Bote Man < <bote_radio at botecomm.com>
>> bote_radio at botecomm.com>:
>>
>>> FreeSWITCH uses UDP by default for SIP signaling. You can change this in
>>> the
>>> SIP_profile I believe.
>>>
>>>
>>> ---
>>> Bote
>>>
>>> FreeSWITCH Docs Janitor
>>> http://freeswitch.org/confluence
>>>
>>>
>>>
>>>
>>> > -----Original Message-----
>>> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-
>>> > users-bounces at lists.freeswitch.org] On Behalf Of Gregor
>>> > Sent: Monday, 01 February, 2016 17:37
>>> > To: freeswitch-users at lists.freeswitch.org
>>> > Subject: [Freeswitch-users] TCP registrations
>>> >
>>> > I think I am missing something.
>>> >
>>> > I would like to configure freeswitch that listens on TCP port for
>>> client
>>> > registrations (internal profile). As I read, freeswitch should do this
>>> by
>>> > default. But freeswitch responses only on UDP protocol. Is there a conf
>>> > setting for specify also tcp for registrations.
>>> >
>>> >
>>> > __________________________________________________________
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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.orghttp://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> --
>
> Saludos / Regards / Cumprimentos,
> Ant?nio silva
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

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 therebel22 at gmail.com  Tue Feb  2 20:20:04 2016
From: therebel22 at gmail.com (Marc S)
Date: Tue, 2 Feb 2016 18:20:04 +0100
Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk
In-Reply-To: <CAMs117X_s+BHbQkPww0jMdC4Jf0z5aV=dy9KAYm3NLc-ojqsww@mail.gmail.com>
References: <CAMs117X_s+BHbQkPww0jMdC4Jf0z5aV=dy9KAYm3NLc-ojqsww@mail.gmail.com>
Message-ID: <CAMs117V-c0hSBJrWBxQkgoxndpVKKMtpbYMbUV_Q5RwC+SSXRA@mail.gmail.com>

Hello,
sorry for asking again :

I hav an asterisk that register on Freeswitch (as a user).

When a call is incoming to FS, FS send it to asterisk : In asterisk, it is
the s extension.

Here my bridge tests

<action application="bridge" data="user/myuserid"/>

=> s extension in asterisk instead of extension

<action application="bridge" data="sofia/mycontext/myextension@
<ASTERISK_IP>"/>

=> Not authenticated in asterisk (because no IP authentication in asterisk)

Have you an idea how to send real extension instead of s extension  ?
Thanks







2016-01-03 10:45 GMT+01:00 Marc S <therebel22 at gmail.com>:

> Hello,
>
> i'm discovering FS. I hav read a lot about users and gateways.
>
> I would like to  FS act as registrar for authenticated SIP trunking.
>
> - Customers IPBX would register with login/password to Freeswitch.
> - Incoming call would be routed to these SIP trunks in dialplan XML.
>
> directory/users does not seem to be the solution because in dialplan,
> destination DID can't be defined, only user id :
>
> <action application="bridge" data="user/myuserid"/>
>
> gateway seems to be designed for SIP trunking to remote SIP gateway, not
> for FS to act as registrar.
>
> Is it possible to FS to act as authenticated SIP trunking registrar ?
>
> Thanks a lot,
> Marc
>
>
>
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From s.safarov at gmail.com  Tue Feb  2 21:16:27 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 2 Feb 2016 21:16:27 +0300
Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk
In-Reply-To: <CAMs117V-c0hSBJrWBxQkgoxndpVKKMtpbYMbUV_Q5RwC+SSXRA@mail.gmail.com>
References: <CAMs117X_s+BHbQkPww0jMdC4Jf0z5aV=dy9KAYm3NLc-ojqsww@mail.gmail.com>
	<CAMs117V-c0hSBJrWBxQkgoxndpVKKMtpbYMbUV_Q5RwC+SSXRA@mail.gmail.com>
Message-ID: <CAHtxdDeKv40QTvRsU6jbewLb-ikOXUVihgpKDCvixiFzuVrUbg@mail.gmail.com>

What is way you planing to use for link DID with user?

Sergey

On Tue, Feb 2, 2016 at 8:20 PM, Marc S <therebel22 at gmail.com> wrote:

> Hello,
> sorry for asking again :
>
> I hav an asterisk that register on Freeswitch (as a user).
>
> When a call is incoming to FS, FS send it to asterisk : In asterisk, it is
> the s extension.
>
> Here my bridge tests
>
> <action application="bridge" data="user/myuserid"/>
>
> => s extension in asterisk instead of extension
>
> <action application="bridge" data="sofia/mycontext/myextension@
> <ASTERISK_IP>"/>
>
> => Not authenticated in asterisk (because no IP authentication in asterisk)
>
> Have you an idea how to send real extension instead of s extension  ?
> Thanks
>
>
>
>
>
>
>
> 2016-01-03 10:45 GMT+01:00 Marc S <therebel22 at gmail.com>:
>
>> Hello,
>>
>> i'm discovering FS. I hav read a lot about users and gateways.
>>
>> I would like to  FS act as registrar for authenticated SIP trunking.
>>
>> - Customers IPBX would register with login/password to Freeswitch.
>> - Incoming call would be routed to these SIP trunks in dialplan XML.
>>
>> directory/users does not seem to be the solution because in dialplan,
>> destination DID can't be defined, only user id :
>>
>> <action application="bridge" data="user/myuserid"/>
>>
>> gateway seems to be designed for SIP trunking to remote SIP gateway, not
>> for FS to act as registrar.
>>
>> Is it possible to FS to act as authenticated SIP trunking registrar ?
>>
>> Thanks a lot,
>> Marc
>>
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 therebel22 at gmail.com  Tue Feb  2 22:03:34 2016
From: therebel22 at gmail.com (Marc S)
Date: Tue, 2 Feb 2016 20:03:34 +0100
Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk
In-Reply-To: <CAHtxdDeKv40QTvRsU6jbewLb-ikOXUVihgpKDCvixiFzuVrUbg@mail.gmail.com>
References: <CAMs117X_s+BHbQkPww0jMdC4Jf0z5aV=dy9KAYm3NLc-ojqsww@mail.gmail.com>
	<CAMs117V-c0hSBJrWBxQkgoxndpVKKMtpbYMbUV_Q5RwC+SSXRA@mail.gmail.com>
	<CAHtxdDeKv40QTvRsU6jbewLb-ikOXUVihgpKDCvixiFzuVrUbg@mail.gmail.com>
Message-ID: <CAMs117X0Y2xwZ6EPMvaPQU9m6tLYswb_8SHte_9bdXDhKa8R8w@mail.gmail.com>

I want to use mod xml curl to generate dynamic dialplan xml like this :

 <extension name="inbound_12345678">
        <condition field="destination_number" expression="^(12345678)$">
           <action application="bridge" data="user/myuserid"/>
        </condition>
</extension>

Thanks



2016-02-02 19:16 GMT+01:00 Sergey Safarov <s.safarov at gmail.com>:

> What is way you planing to use for link DID with user?
>
> Sergey
>
> On Tue, Feb 2, 2016 at 8:20 PM, Marc S <therebel22 at gmail.com> wrote:
>
>> Hello,
>> sorry for asking again :
>>
>> I hav an asterisk that register on Freeswitch (as a user).
>>
>> When a call is incoming to FS, FS send it to asterisk : In asterisk, it
>> is the s extension.
>>
>> Here my bridge tests
>>
>> <action application="bridge" data="user/myuserid"/>
>>
>> => s extension in asterisk instead of extension
>>
>> <action application="bridge" data="sofia/mycontext/myextension@
>> <ASTERISK_IP>"/>
>>
>> => Not authenticated in asterisk (because no IP authentication in
>> asterisk)
>>
>> Have you an idea how to send real extension instead of s extension  ?
>> Thanks
>>
>>
>>
>>
>>
>>
>>
>> 2016-01-03 10:45 GMT+01:00 Marc S <therebel22 at gmail.com>:
>>
>>> Hello,
>>>
>>> i'm discovering FS. I hav read a lot about users and gateways.
>>>
>>> I would like to  FS act as registrar for authenticated SIP trunking.
>>>
>>> - Customers IPBX would register with login/password to Freeswitch.
>>> - Incoming call would be routed to these SIP trunks in dialplan XML.
>>>
>>> directory/users does not seem to be the solution because in dialplan,
>>> destination DID can't be defined, only user id :
>>>
>>> <action application="bridge" data="user/myuserid"/>
>>>
>>> gateway seems to be designed for SIP trunking to remote SIP gateway, not
>>> for FS to act as registrar.
>>>
>>> Is it possible to FS to act as authenticated SIP trunking registrar ?
>>>
>>> Thanks a lot,
>>> Marc
>>>
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 omortimer at gmail.com  Tue Feb  2 22:23:16 2016
From: omortimer at gmail.com (Oz Mortimer)
Date: Tue, 2 Feb 2016 19:23:16 +0000
Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk
In-Reply-To: <CAMs117X0Y2xwZ6EPMvaPQU9m6tLYswb_8SHte_9bdXDhKa8R8w@mail.gmail.com>
References: <CAMs117X_s+BHbQkPww0jMdC4Jf0z5aV=dy9KAYm3NLc-ojqsww@mail.gmail.com>
	<CAMs117V-c0hSBJrWBxQkgoxndpVKKMtpbYMbUV_Q5RwC+SSXRA@mail.gmail.com>
	<CAHtxdDeKv40QTvRsU6jbewLb-ikOXUVihgpKDCvixiFzuVrUbg@mail.gmail.com>
	<CAMs117X0Y2xwZ6EPMvaPQU9m6tLYswb_8SHte_9bdXDhKa8R8w@mail.gmail.com>
Message-ID: <490EAD37-3FE1-4A94-8166-499B461F9C53@gmail.com>

Setup username / password authentication on asterisk and set the corresponding user & pass in your freeswitch gateway. 
I'm sure a google for "asterisk username authentication" and "freeswitch gateway username" will give you plenty of examples - you will want freeswitch to bridge to gateway - again Google "freeswitch bridge gateway".
Google is your friend.. 


> On 2 Feb 2016, at 19:03, Marc S <therebel22 at gmail.com> wrote:
> 
> I want to use mod xml curl to generate dynamic dialplan xml like this :
> 
>  <extension name="inbound_12345678">
>         <condition field="destination_number" expression="^(12345678)$">
>            <action application="bridge" data="user/myuserid"/>
>         </condition>
> </extension>
> 
> Thanks
> 
> 
> 
> 2016-02-02 19:16 GMT+01:00 Sergey Safarov <s.safarov at gmail.com>:
>> What is way you planing to use for link DID with user?
>> 
>> Sergey
>> 
>>> On Tue, Feb 2, 2016 at 8:20 PM, Marc S <therebel22 at gmail.com> wrote:
>>> Hello,
>>> sorry for asking again :
>>> 
>>> I hav an asterisk that register on Freeswitch (as a user). 
>>> 
>>> When a call is incoming to FS, FS send it to asterisk : In asterisk, it is the s extension.
>>> 
>>> Here my bridge tests
>>> 
>>> <action application="bridge" data="user/myuserid"/> 
>>> 
>>> => s extension in asterisk instead of extension
>>> 
>>> <action application="bridge" data="sofia/mycontext/myextension@<ASTERISK_IP>"/>
>>> 
>>> => Not authenticated in asterisk (because no IP authentication in asterisk)
>>> 
>>> Have you an idea how to send real extension instead of s extension  ?
>>> Thanks
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 2016-01-03 10:45 GMT+01:00 Marc S <therebel22 at gmail.com>:
>>>> Hello,
>>>> 
>>>> i'm discovering FS. I hav read a lot about users and gateways.
>>>> 
>>>> I would like to  FS act as registrar for authenticated SIP trunking.
>>>> 
>>>> - Customers IPBX would register with login/password to Freeswitch. 
>>>> - Incoming call would be routed to these SIP trunks in dialplan XML.
>>>> 
>>>> directory/users does not seem to be the solution because in dialplan, destination DID can't be defined, only user id :
>>>> 
>>>> <action application="bridge" data="user/myuserid"/>
>>>> 
>>>> gateway seems to be designed for SIP trunking to remote SIP gateway, not for FS to act as registrar.
>>>> 
>>>> Is it possible to FS to act as authenticated SIP trunking registrar ?
>>>> 
>>>> Thanks a lot,
>>>> Marc
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 therebel22 at gmail.com  Tue Feb  2 22:28:11 2016
From: therebel22 at gmail.com (Marc S)
Date: Tue, 2 Feb 2016 20:28:11 +0100
Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk
In-Reply-To: <490EAD37-3FE1-4A94-8166-499B461F9C53@gmail.com>
References: <CAMs117X_s+BHbQkPww0jMdC4Jf0z5aV=dy9KAYm3NLc-ojqsww@mail.gmail.com>
	<CAMs117V-c0hSBJrWBxQkgoxndpVKKMtpbYMbUV_Q5RwC+SSXRA@mail.gmail.com>
	<CAHtxdDeKv40QTvRsU6jbewLb-ikOXUVihgpKDCvixiFzuVrUbg@mail.gmail.com>
	<CAMs117X0Y2xwZ6EPMvaPQU9m6tLYswb_8SHte_9bdXDhKa8R8w@mail.gmail.com>
	<490EAD37-3FE1-4A94-8166-499B461F9C53@gmail.com>
Message-ID: <CAMs117W+g2cFaQ-N0ULKMS-EDywazOsvfakBaQ4Hzv8VZ4LGnw@mail.gmail.com>

It seems that "freeswitch gateway username" return all results about setup
username and password on FS to register against external SIP gateway, am i
wrong ?

2016-02-02 20:23 GMT+01:00 Oz Mortimer <omortimer at gmail.com>:

> Setup username / password authentication on asterisk and set the
> corresponding user & pass in your freeswitch gateway.
> I'm sure a google for "asterisk username authentication" and "freeswitch
> gateway username" will give you plenty of examples - you will want
> freeswitch to bridge to gateway - again Google "freeswitch bridge gateway".
> Google is your friend..
>
>
> On 2 Feb 2016, at 19:03, Marc S <therebel22 at gmail.com> wrote:
>
> I want to use mod xml curl to generate dynamic dialplan xml like this :
>
>  <extension name="inbound_12345678">
>         <condition field="destination_number" expression="^(12345678)$">
>            <action application="bridge" data="user/myuserid"/>
>         </condition>
> </extension>
>
> Thanks
>
>
>
> 2016-02-02 19:16 GMT+01:00 Sergey Safarov <s.safarov at gmail.com>:
>
>> What is way you planing to use for link DID with user?
>>
>> Sergey
>>
>> On Tue, Feb 2, 2016 at 8:20 PM, Marc S <therebel22 at gmail.com> wrote:
>>
>>> Hello,
>>> sorry for asking again :
>>>
>>> I hav an asterisk that register on Freeswitch (as a user).
>>>
>>> When a call is incoming to FS, FS send it to asterisk : In asterisk, it
>>> is the s extension.
>>>
>>> Here my bridge tests
>>>
>>> <action application="bridge" data="user/myuserid"/>
>>>
>>> => s extension in asterisk instead of extension
>>>
>>> <action application="bridge" data="sofia/mycontext/myextension@
>>> <ASTERISK_IP>"/>
>>>
>>> => Not authenticated in asterisk (because no IP authentication in
>>> asterisk)
>>>
>>> Have you an idea how to send real extension instead of s extension  ?
>>> Thanks
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> 2016-01-03 10:45 GMT+01:00 Marc S <therebel22 at gmail.com>:
>>>
>>>> Hello,
>>>>
>>>> i'm discovering FS. I hav read a lot about users and gateways.
>>>>
>>>> I would like to  FS act as registrar for authenticated SIP trunking.
>>>>
>>>> - Customers IPBX would register with login/password to Freeswitch.
>>>> - Incoming call would be routed to these SIP trunks in dialplan XML.
>>>>
>>>> directory/users does not seem to be the solution because in dialplan,
>>>> destination DID can't be defined, only user id :
>>>>
>>>> <action application="bridge" data="user/myuserid"/>
>>>>
>>>> gateway seems to be designed for SIP trunking to remote SIP gateway,
>>>> not for FS to act as registrar.
>>>>
>>>> Is it possible to FS to act as authenticated SIP trunking registrar ?
>>>>
>>>> Thanks a lot,
>>>> Marc
>>>>
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 blessendor at gmail.com  Tue Feb  2 22:57:12 2016
From: blessendor at gmail.com (Alexandr Usov)
Date: Tue, 2 Feb 2016 21:57:12 +0200
Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk
In-Reply-To: <CAMs117W+g2cFaQ-N0ULKMS-EDywazOsvfakBaQ4Hzv8VZ4LGnw@mail.gmail.com>
References: <CAMs117X_s+BHbQkPww0jMdC4Jf0z5aV=dy9KAYm3NLc-ojqsww@mail.gmail.com>
	<CAMs117V-c0hSBJrWBxQkgoxndpVKKMtpbYMbUV_Q5RwC+SSXRA@mail.gmail.com>
	<CAHtxdDeKv40QTvRsU6jbewLb-ikOXUVihgpKDCvixiFzuVrUbg@mail.gmail.com>
	<CAMs117X0Y2xwZ6EPMvaPQU9m6tLYswb_8SHte_9bdXDhKa8R8w@mail.gmail.com>
	<490EAD37-3FE1-4A94-8166-499B461F9C53@gmail.com>
	<CAMs117W+g2cFaQ-N0ULKMS-EDywazOsvfakBaQ4Hzv8VZ4LGnw@mail.gmail.com>
Message-ID: <CAEaKXwD32DyhS-tJCJ8BPExMLwRqG4GqREZWhdi8fJ9i6+sWPg@mail.gmail.com>

You must have in Asterisk configs something as in my examples.

;;Register on freeswitch if your asterisk behind dynamic IP or NAT, etc.
With static IP of both FS and Asterisk you not need to do register from
asterisk.
register => asterisk2fs:strongpassword at ip_of_freeswitch/freeswitch


;; type=friend means that this peer can be used for outbound and inbound
calls, so we don't need to create two peer settings block ( [freeswitch-in]
and [freeswitch-out] )

[freeswitch] ;; you maybe want to use here your public DID number, as well
as in username/fromuser settings
type=friend
username=asterisk2fs
fromuser=asterisk2fs
fromdomain=ip_of_freeswitch
host=ip_of_freeswitch
context=from-freeswitch
secret=strongpassword
insecure=port,invite
qualify=yes
port=5060


Your context [from-freeswitch] must have an extension, named as
'freeswitch' (or your DID number) for incoming calls operations.




2016-02-02 21:28 GMT+02:00 Marc S <therebel22 at gmail.com>:

> It seems that "freeswitch gateway username" return all results about setup
> username and password on FS to register against external SIP gateway, am i
> wrong ?
>
> 2016-02-02 20:23 GMT+01:00 Oz Mortimer <omortimer at gmail.com>:
>
>> Setup username / password authentication on asterisk and set the
>> corresponding user & pass in your freeswitch gateway.
>> I'm sure a google for "asterisk username authentication" and "freeswitch
>> gateway username" will give you plenty of examples - you will want
>> freeswitch to bridge to gateway - again Google "freeswitch bridge gateway".
>> Google is your friend..
>>
>>
>> On 2 Feb 2016, at 19:03, Marc S <therebel22 at gmail.com> wrote:
>>
>> I want to use mod xml curl to generate dynamic dialplan xml like this :
>>
>>  <extension name="inbound_12345678">
>>         <condition field="destination_number" expression="^(12345678)$">
>>            <action application="bridge" data="user/myuserid"/>
>>         </condition>
>> </extension>
>>
>> Thanks
>>
>>
>>
>> 2016-02-02 19:16 GMT+01:00 Sergey Safarov <s.safarov at gmail.com>:
>>
>>> What is way you planing to use for link DID with user?
>>>
>>> Sergey
>>>
>>> On Tue, Feb 2, 2016 at 8:20 PM, Marc S <therebel22 at gmail.com> wrote:
>>>
>>>> Hello,
>>>> sorry for asking again :
>>>>
>>>> I hav an asterisk that register on Freeswitch (as a user).
>>>>
>>>> When a call is incoming to FS, FS send it to asterisk : In asterisk, it
>>>> is the s extension.
>>>>
>>>> Here my bridge tests
>>>>
>>>> <action application="bridge" data="user/myuserid"/>
>>>>
>>>> => s extension in asterisk instead of extension
>>>>
>>>> <action application="bridge" data="sofia/mycontext/myextension@
>>>> <ASTERISK_IP>"/>
>>>>
>>>> => Not authenticated in asterisk (because no IP authentication in
>>>> asterisk)
>>>>
>>>> Have you an idea how to send real extension instead of s extension  ?
>>>> Thanks
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> 2016-01-03 10:45 GMT+01:00 Marc S <therebel22 at gmail.com>:
>>>>
>>>>> Hello,
>>>>>
>>>>> i'm discovering FS. I hav read a lot about users and gateways.
>>>>>
>>>>> I would like to  FS act as registrar for authenticated SIP trunking.
>>>>>
>>>>> - Customers IPBX would register with login/password to Freeswitch.
>>>>> - Incoming call would be routed to these SIP trunks in dialplan XML.
>>>>>
>>>>> directory/users does not seem to be the solution because in dialplan,
>>>>> destination DID can't be defined, only user id :
>>>>>
>>>>> <action application="bridge" data="user/myuserid"/>
>>>>>
>>>>> gateway seems to be designed for SIP trunking to remote SIP gateway,
>>>>> not for FS to act as registrar.
>>>>>
>>>>> Is it possible to FS to act as authenticated SIP trunking registrar ?
>>>>>
>>>>> Thanks a lot,
>>>>> Marc
>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 therebel22 at gmail.com  Tue Feb  2 23:06:01 2016
From: therebel22 at gmail.com (Marc S)
Date: Tue, 2 Feb 2016 21:06:01 +0100
Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk
In-Reply-To: <CAEaKXwD32DyhS-tJCJ8BPExMLwRqG4GqREZWhdi8fJ9i6+sWPg@mail.gmail.com>
References: <CAMs117X_s+BHbQkPww0jMdC4Jf0z5aV=dy9KAYm3NLc-ojqsww@mail.gmail.com>
	<CAMs117V-c0hSBJrWBxQkgoxndpVKKMtpbYMbUV_Q5RwC+SSXRA@mail.gmail.com>
	<CAHtxdDeKv40QTvRsU6jbewLb-ikOXUVihgpKDCvixiFzuVrUbg@mail.gmail.com>
	<CAMs117X0Y2xwZ6EPMvaPQU9m6tLYswb_8SHte_9bdXDhKa8R8w@mail.gmail.com>
	<490EAD37-3FE1-4A94-8166-499B461F9C53@gmail.com>
	<CAMs117W+g2cFaQ-N0ULKMS-EDywazOsvfakBaQ4Hzv8VZ4LGnw@mail.gmail.com>
	<CAEaKXwD32DyhS-tJCJ8BPExMLwRqG4GqREZWhdi8fJ9i6+sWPg@mail.gmail.com>
Message-ID: <CAMs117XCM9bv+Mq9feQ1BjyBRLBkkc9OcbAXNDQG4AO755cbUQ@mail.gmail.com>

Thanks, Asterisk is already registred to FS.

But when incoming call to FS is bridged to registered Asterisk :

<action application="bridge" data="user/myuserid"/>

FS send SIP Message to asterisk :

INVITE s@<asterisk_ip>

instead of :

INVITE 12345678@<asterisk_ip>

I would like to get 12345678 in asterisk..




2016-02-02 20:57 GMT+01:00 Alexandr Usov <blessendor at gmail.com>:

> You must have in Asterisk configs something as in my examples.
>
> ;;Register on freeswitch if your asterisk behind dynamic IP or NAT, etc.
> With static IP of both FS and Asterisk you not need to do register from
> asterisk.
> register => asterisk2fs:strongpassword at ip_of_freeswitch/freeswitch
>
>
> ;; type=friend means that this peer can be used for outbound and inbound
> calls, so we don't need to create two peer settings block ( [freeswitch-in]
> and [freeswitch-out] )
>
> [freeswitch] ;; you maybe want to use here your public DID number, as well
> as in username/fromuser settings
> type=friend
> username=asterisk2fs
> fromuser=asterisk2fs
> fromdomain=ip_of_freeswitch
> host=ip_of_freeswitch
> context=from-freeswitch
> secret=strongpassword
> insecure=port,invite
> qualify=yes
> port=5060
>
>
> Your context [from-freeswitch] must have an extension, named as
> 'freeswitch' (or your DID number) for incoming calls operations.
>
>
>
>
> 2016-02-02 21:28 GMT+02:00 Marc S <therebel22 at gmail.com>:
>
>> It seems that "freeswitch gateway username" return all results about
>> setup username and password on FS to register against external SIP gateway,
>> am i wrong ?
>>
>> 2016-02-02 20:23 GMT+01:00 Oz Mortimer <omortimer at gmail.com>:
>>
>>> Setup username / password authentication on asterisk and set the
>>> corresponding user & pass in your freeswitch gateway.
>>> I'm sure a google for "asterisk username authentication" and "freeswitch
>>> gateway username" will give you plenty of examples - you will want
>>> freeswitch to bridge to gateway - again Google "freeswitch bridge gateway".
>>> Google is your friend..
>>>
>>>
>>> On 2 Feb 2016, at 19:03, Marc S <therebel22 at gmail.com> wrote:
>>>
>>> I want to use mod xml curl to generate dynamic dialplan xml like this :
>>>
>>>  <extension name="inbound_12345678">
>>>         <condition field="destination_number" expression="^(12345678)$">
>>>            <action application="bridge" data="user/myuserid"/>
>>>         </condition>
>>> </extension>
>>>
>>> Thanks
>>>
>>>
>>>
>>> 2016-02-02 19:16 GMT+01:00 Sergey Safarov <s.safarov at gmail.com>:
>>>
>>>> What is way you planing to use for link DID with user?
>>>>
>>>> Sergey
>>>>
>>>> On Tue, Feb 2, 2016 at 8:20 PM, Marc S <therebel22 at gmail.com> wrote:
>>>>
>>>>> Hello,
>>>>> sorry for asking again :
>>>>>
>>>>> I hav an asterisk that register on Freeswitch (as a user).
>>>>>
>>>>> When a call is incoming to FS, FS send it to asterisk : In asterisk,
>>>>> it is the s extension.
>>>>>
>>>>> Here my bridge tests
>>>>>
>>>>> <action application="bridge" data="user/myuserid"/>
>>>>>
>>>>> => s extension in asterisk instead of extension
>>>>>
>>>>> <action application="bridge" data="sofia/mycontext/myextension@
>>>>> <ASTERISK_IP>"/>
>>>>>
>>>>> => Not authenticated in asterisk (because no IP authentication in
>>>>> asterisk)
>>>>>
>>>>> Have you an idea how to send real extension instead of s extension  ?
>>>>> Thanks
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> 2016-01-03 10:45 GMT+01:00 Marc S <therebel22 at gmail.com>:
>>>>>
>>>>>> Hello,
>>>>>>
>>>>>> i'm discovering FS. I hav read a lot about users and gateways.
>>>>>>
>>>>>> I would like to  FS act as registrar for authenticated SIP trunking.
>>>>>>
>>>>>> - Customers IPBX would register with login/password to Freeswitch.
>>>>>> - Incoming call would be routed to these SIP trunks in dialplan XML.
>>>>>>
>>>>>> directory/users does not seem to be the solution because in dialplan,
>>>>>> destination DID can't be defined, only user id :
>>>>>>
>>>>>> <action application="bridge" data="user/myuserid"/>
>>>>>>
>>>>>> gateway seems to be designed for SIP trunking to remote SIP gateway,
>>>>>> not for FS to act as registrar.
>>>>>>
>>>>>> Is it possible to FS to act as authenticated SIP trunking registrar ?
>>>>>>
>>>>>> Thanks a lot,
>>>>>> Marc
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 blessendor at gmail.com  Tue Feb  2 23:10:25 2016
From: blessendor at gmail.com (Alexandr Usov)
Date: Tue, 2 Feb 2016 22:10:25 +0200
Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk
In-Reply-To: <CAMs117XCM9bv+Mq9feQ1BjyBRLBkkc9OcbAXNDQG4AO755cbUQ@mail.gmail.com>
References: <CAMs117X_s+BHbQkPww0jMdC4Jf0z5aV=dy9KAYm3NLc-ojqsww@mail.gmail.com>
	<CAMs117V-c0hSBJrWBxQkgoxndpVKKMtpbYMbUV_Q5RwC+SSXRA@mail.gmail.com>
	<CAHtxdDeKv40QTvRsU6jbewLb-ikOXUVihgpKDCvixiFzuVrUbg@mail.gmail.com>
	<CAMs117X0Y2xwZ6EPMvaPQU9m6tLYswb_8SHte_9bdXDhKa8R8w@mail.gmail.com>
	<490EAD37-3FE1-4A94-8166-499B461F9C53@gmail.com>
	<CAMs117W+g2cFaQ-N0ULKMS-EDywazOsvfakBaQ4Hzv8VZ4LGnw@mail.gmail.com>
	<CAEaKXwD32DyhS-tJCJ8BPExMLwRqG4GqREZWhdi8fJ9i6+sWPg@mail.gmail.com>
	<CAMs117XCM9bv+Mq9feQ1BjyBRLBkkc9OcbAXNDQG4AO755cbUQ@mail.gmail.com>
Message-ID: <CAEaKXwBU_LrBt-CrC962Z53LaziJA4g4kPj70vFSMKY52RRuJA@mail.gmail.com>

Post you register and incoming sip peer settings from yor asterisk if you
want to got further help.


2016-02-02 22:06 GMT+02:00 Marc S <therebel22 at gmail.com>:

> Thanks, Asterisk is already registred to FS.
>
> But when incoming call to FS is bridged to registered Asterisk :
>
> <action application="bridge" data="user/myuserid"/>
>
> FS send SIP Message to asterisk :
>
> INVITE s@<asterisk_ip>
>
> instead of :
>
> INVITE 12345678@<asterisk_ip>
>
> I would like to get 12345678 in asterisk..
>
>
>
>
> 2016-02-02 20:57 GMT+01:00 Alexandr Usov <blessendor at gmail.com>:
>
>> You must have in Asterisk configs something as in my examples.
>>
>> ;;Register on freeswitch if your asterisk behind dynamic IP or NAT, etc.
>> With static IP of both FS and Asterisk you not need to do register from
>> asterisk.
>> register => asterisk2fs:strongpassword at ip_of_freeswitch/freeswitch
>>
>>
>> ;; type=friend means that this peer can be used for outbound and inbound
>> calls, so we don't need to create two peer settings block ( [freeswitch-in]
>> and [freeswitch-out] )
>>
>> [freeswitch] ;; you maybe want to use here your public DID number, as
>> well as in username/fromuser settings
>> type=friend
>> username=asterisk2fs
>> fromuser=asterisk2fs
>> fromdomain=ip_of_freeswitch
>> host=ip_of_freeswitch
>> context=from-freeswitch
>> secret=strongpassword
>> insecure=port,invite
>> qualify=yes
>> port=5060
>>
>>
>> Your context [from-freeswitch] must have an extension, named as
>> 'freeswitch' (or your DID number) for incoming calls operations.
>>
>>
>>
>>
>> 2016-02-02 21:28 GMT+02:00 Marc S <therebel22 at gmail.com>:
>>
>>> It seems that "freeswitch gateway username" return all results about
>>> setup username and password on FS to register against external SIP gateway,
>>> am i wrong ?
>>>
>>> 2016-02-02 20:23 GMT+01:00 Oz Mortimer <omortimer at gmail.com>:
>>>
>>>> Setup username / password authentication on asterisk and set the
>>>> corresponding user & pass in your freeswitch gateway.
>>>> I'm sure a google for "asterisk username authentication" and
>>>> "freeswitch gateway username" will give you plenty of examples - you will
>>>> want freeswitch to bridge to gateway - again Google "freeswitch bridge
>>>> gateway".
>>>> Google is your friend..
>>>>
>>>>
>>>> On 2 Feb 2016, at 19:03, Marc S <therebel22 at gmail.com> wrote:
>>>>
>>>> I want to use mod xml curl to generate dynamic dialplan xml like this :
>>>>
>>>>  <extension name="inbound_12345678">
>>>>         <condition field="destination_number" expression="^(12345678)$">
>>>>            <action application="bridge" data="user/myuserid"/>
>>>>         </condition>
>>>> </extension>
>>>>
>>>> Thanks
>>>>
>>>>
>>>>
>>>> 2016-02-02 19:16 GMT+01:00 Sergey Safarov <s.safarov at gmail.com>:
>>>>
>>>>> What is way you planing to use for link DID with user?
>>>>>
>>>>> Sergey
>>>>>
>>>>> On Tue, Feb 2, 2016 at 8:20 PM, Marc S <therebel22 at gmail.com> wrote:
>>>>>
>>>>>> Hello,
>>>>>> sorry for asking again :
>>>>>>
>>>>>> I hav an asterisk that register on Freeswitch (as a user).
>>>>>>
>>>>>> When a call is incoming to FS, FS send it to asterisk : In asterisk,
>>>>>> it is the s extension.
>>>>>>
>>>>>> Here my bridge tests
>>>>>>
>>>>>> <action application="bridge" data="user/myuserid"/>
>>>>>>
>>>>>> => s extension in asterisk instead of extension
>>>>>>
>>>>>> <action application="bridge" data="sofia/mycontext/myextension@
>>>>>> <ASTERISK_IP>"/>
>>>>>>
>>>>>> => Not authenticated in asterisk (because no IP authentication in
>>>>>> asterisk)
>>>>>>
>>>>>> Have you an idea how to send real extension instead of s extension  ?
>>>>>> Thanks
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> 2016-01-03 10:45 GMT+01:00 Marc S <therebel22 at gmail.com>:
>>>>>>
>>>>>>> Hello,
>>>>>>>
>>>>>>> i'm discovering FS. I hav read a lot about users and gateways.
>>>>>>>
>>>>>>> I would like to  FS act as registrar for authenticated SIP trunking.
>>>>>>>
>>>>>>> - Customers IPBX would register with login/password to Freeswitch.
>>>>>>> - Incoming call would be routed to these SIP trunks in dialplan XML.
>>>>>>>
>>>>>>> directory/users does not seem to be the solution because in
>>>>>>> dialplan, destination DID can't be defined, only user id :
>>>>>>>
>>>>>>> <action application="bridge" data="user/myuserid"/>
>>>>>>>
>>>>>>> gateway seems to be designed for SIP trunking to remote SIP gateway,
>>>>>>> not for FS to act as registrar.
>>>>>>>
>>>>>>> Is it possible to FS to act as authenticated SIP trunking registrar ?
>>>>>>>
>>>>>>> Thanks a lot,
>>>>>>> Marc
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://
>>>> lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 sos at sokhapkin.dyndns.org  Tue Feb  2 23:13:50 2016
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 02 Feb 2016 15:13:50 -0500
Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk
In-Reply-To: <CAMs117XCM9bv+Mq9feQ1BjyBRLBkkc9OcbAXNDQG4AO755cbUQ@mail.gmail.com>
References: <CAMs117X_s+BHbQkPww0jMdC4Jf0z5aV=dy9KAYm3NLc-ojqsww@mail.gmail.com>
	<CAEaKXwD32DyhS-tJCJ8BPExMLwRqG4GqREZWhdi8fJ9i6+sWPg@mail.gmail.com>
	<CAMs117XCM9bv+Mq9feQ1BjyBRLBkkc9OcbAXNDQG4AO755cbUQ@mail.gmail.com>
Message-ID: <1926695.hkLRVMayM7@sos>

Look at REGISTER messages from asterisk. They have header 

Contact: s@<asterisk_ip>

You get what you ask for :-)

On Tuesday 02 February 2016 21:06:01 Marc S wrote:
> Thanks, Asterisk is already registred to FS.
> 
> But when incoming call to FS is bridged to registered Asterisk :
> 
> <action application="bridge" data="user/myuserid"/>
> 
> FS send SIP Message to asterisk :
> 
> INVITE s@<asterisk_ip>
> 
> instead of :
> 
> INVITE 12345678@<asterisk_ip>
> 
> I would like to get 12345678 in asterisk..
> 
> 2016-02-02 20:57 GMT+01:00 Alexandr Usov <blessendor at gmail.com>:
> > You must have in Asterisk configs something as in my examples.
> > 
> > ;;Register on freeswitch if your asterisk behind dynamic IP or NAT, etc.
> > With static IP of both FS and Asterisk you not need to do register from
> > asterisk.
> > register => asterisk2fs:strongpassword at ip_of_freeswitch/freeswitch
> > 
> > 
> > ;; type=friend means that this peer can be used for outbound and inbound
> > calls, so we don't need to create two peer settings block (
> > [freeswitch-in]
> > and [freeswitch-out] )
> > 
> > [freeswitch] ;; you maybe want to use here your public DID number, as well
> > as in username/fromuser settings
> > type=friend
> > username=asterisk2fs
> > fromuser=asterisk2fs
> > fromdomain=ip_of_freeswitch
> > host=ip_of_freeswitch
> > context=from-freeswitch
> > secret=strongpassword
> > insecure=port,invite
> > qualify=yes
> > port=5060
> > 
> > 
> > Your context [from-freeswitch] must have an extension, named as
> > 'freeswitch' (or your DID number) for incoming calls operations.
> > 
> > 2016-02-02 21:28 GMT+02:00 Marc S <therebel22 at gmail.com>:
> >> It seems that "freeswitch gateway username" return all results about
> >> setup username and password on FS to register against external SIP
> >> gateway,
> >> am i wrong ?
> >> 
> >> 2016-02-02 20:23 GMT+01:00 Oz Mortimer <omortimer at gmail.com>:
> >>> Setup username / password authentication on asterisk and set the
> >>> corresponding user & pass in your freeswitch gateway.
> >>> I'm sure a google for "asterisk username authentication" and "freeswitch
> >>> gateway username" will give you plenty of examples - you will want
> >>> freeswitch to bridge to gateway - again Google "freeswitch bridge
> >>> gateway".
> >>> Google is your friend..
> >>> 
> >>> 
> >>> On 2 Feb 2016, at 19:03, Marc S <therebel22 at gmail.com> wrote:
> >>> 
> >>> I want to use mod xml curl to generate dynamic dialplan xml like this :
> >>>  <extension name="inbound_12345678">
> >>>  
> >>>         <condition field="destination_number" expression="^(12345678)$">
> >>>         
> >>>            <action application="bridge" data="user/myuserid"/>
> >>>         
> >>>         </condition>
> >>> 
> >>> </extension>
> >>> 
> >>> Thanks
> >>> 
> >>> 2016-02-02 19:16 GMT+01:00 Sergey Safarov <s.safarov at gmail.com>:
> >>>> What is way you planing to use for link DID with user?
> >>>> 
> >>>> Sergey
> >>>> 
> >>>> On Tue, Feb 2, 2016 at 8:20 PM, Marc S <therebel22 at gmail.com> wrote:
> >>>>> Hello,
> >>>>> sorry for asking again :
> >>>>> 
> >>>>> I hav an asterisk that register on Freeswitch (as a user).
> >>>>> 
> >>>>> When a call is incoming to FS, FS send it to asterisk : In asterisk,
> >>>>> it is the s extension.
> >>>>> 
> >>>>> Here my bridge tests
> >>>>> 
> >>>>> <action application="bridge" data="user/myuserid"/>
> >>>>> 
> >>>>> => s extension in asterisk instead of extension
> >>>>> 
> >>>>> <action application="bridge" data="sofia/mycontext/myextension@
> >>>>> <ASTERISK_IP>"/>
> >>>>> 
> >>>>> => Not authenticated in asterisk (because no IP authentication in
> >>>>> asterisk)
> >>>>> 
> >>>>> Have you an idea how to send real extension instead of s extension  ?
> >>>>> Thanks
> >>>>> 
> >>>>> 2016-01-03 10:45 GMT+01:00 Marc S <therebel22 at gmail.com>:
> >>>>>> Hello,
> >>>>>> 
> >>>>>> i'm discovering FS. I hav read a lot about users and gateways.
> >>>>>> 
> >>>>>> I would like to  FS act as registrar for authenticated SIP trunking.
> >>>>>> 
> >>>>>> - Customers IPBX would register with login/password to Freeswitch.
> >>>>>> - Incoming call would be routed to these SIP trunks in dialplan XML.
> >>>>>> 
> >>>>>> directory/users does not seem to be the solution because in dialplan,
> >>>>>> destination DID can't be defined, only user id :
> >>>>>> 
> >>>>>> <action application="bridge" data="user/myuserid"/>
> >>>>>> 
> >>>>>> gateway seems to be designed for SIP trunking to remote SIP gateway,
> >>>>>> not for FS to act as registrar.
> >>>>>> 
> >>>>>> Is it possible to FS to act as authenticated SIP trunking registrar ?
> >>>>>> 
> >>>>>> Thanks a lot,
> >>>>>> Marc
> >>>>> 
> >>>>> ______________________________________________________________________
> >>>>> ___
> >>>>> Professional FreeSWITCH Consulting Services:
> >>>>> consulting at freeswitch.org
> >>>>> http://www.freeswitchsolutions.com
> >>>>> 
> >>>>> Official FreeSWITCH Sites
> >>>>> http://www.freeswitch.org
> >>>>> http://confluence.freeswitch.org
> >>>>> http://www.cluecon.com
> >>>>> 
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>> UNSUBSCRIBE:
> >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>> 
> >>>> _______________________________________________________________________
> >>>> __
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>> 
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://confluence.freeswitch.org
> >>>> http://www.cluecon.com
> >>>> 
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>> 
> >>> ________________________________________________________________________
> >>> _
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>> 
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://confluence.freeswitch.org
> >>> http://www.cluecon.com
> >>> 
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>> 
> >>> 
> >>> ________________________________________________________________________
> >>> _
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>> 
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://confluence.freeswitch.org
> >>> http://www.cluecon.com
> >>> 
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >> 
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >> 
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >> 
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> > 
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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 brian at freeswitch.org  Tue Feb  2 23:14:31 2016
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Feb 2016 14:14:31 -0600
Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk
In-Reply-To: <CAMs117XCM9bv+Mq9feQ1BjyBRLBkkc9OcbAXNDQG4AO755cbUQ@mail.gmail.com>
References: <CAMs117X_s+BHbQkPww0jMdC4Jf0z5aV=dy9KAYm3NLc-ojqsww@mail.gmail.com>
	<CAMs117V-c0hSBJrWBxQkgoxndpVKKMtpbYMbUV_Q5RwC+SSXRA@mail.gmail.com>
	<CAHtxdDeKv40QTvRsU6jbewLb-ikOXUVihgpKDCvixiFzuVrUbg@mail.gmail.com>
	<CAMs117X0Y2xwZ6EPMvaPQU9m6tLYswb_8SHte_9bdXDhKa8R8w@mail.gmail.com>
	<490EAD37-3FE1-4A94-8166-499B461F9C53@gmail.com>
	<CAMs117W+g2cFaQ-N0ULKMS-EDywazOsvfakBaQ4Hzv8VZ4LGnw@mail.gmail.com>
	<CAEaKXwD32DyhS-tJCJ8BPExMLwRqG4GqREZWhdi8fJ9i6+sWPg@mail.gmail.com>
	<CAMs117XCM9bv+Mq9feQ1BjyBRLBkkc9OcbAXNDQG4AO755cbUQ@mail.gmail.com>
Message-ID: <CAEJMVkDJ19zs0o6cZK0dj8z_1W8U+LXYs966g7AmKGKrb4QLpg@mail.gmail.com>

This is because Asterisk will register as s@ on the contact when it
registers to FreeSWITCH, so you can fix this by fixing your register line
in sip.conf on Asterisk to register as 12345678 vs the default s

If I recall its this format:

register => user[:secret[:authuser]]@host[:port][/extension]


Thanks,

On Tue, Feb 2, 2016 at 2:06 PM, Marc S <therebel22 at gmail.com> wrote:

> Thanks, Asterisk is already registred to FS.
>
> But when incoming call to FS is bridged to registered Asterisk :
>
> <action application="bridge" data="user/myuserid"/>
>
> FS send SIP Message to asterisk :
>
> INVITE s@<asterisk_ip>
>
> instead of :
>
> INVITE 12345678@<asterisk_ip>
>
> I would like to get 12345678 in asterisk..
>
>
>
>
> 2016-02-02 20:57 GMT+01:00 Alexandr Usov <blessendor at gmail.com>:
>
>> You must have in Asterisk configs something as in my examples.
>>
>> ;;Register on freeswitch if your asterisk behind dynamic IP or NAT, etc.
>> With static IP of both FS and Asterisk you not need to do register from
>> asterisk.
>> register => asterisk2fs:strongpassword at ip_of_freeswitch/freeswitch
>>
>>
>> ;; type=friend means that this peer can be used for outbound and inbound
>> calls, so we don't need to create two peer settings block ( [freeswitch-in]
>> and [freeswitch-out] )
>>
>> [freeswitch] ;; you maybe want to use here your public DID number, as
>> well as in username/fromuser settings
>> type=friend
>> username=asterisk2fs
>> fromuser=asterisk2fs
>> fromdomain=ip_of_freeswitch
>> host=ip_of_freeswitch
>> context=from-freeswitch
>> secret=strongpassword
>> insecure=port,invite
>> qualify=yes
>> port=5060
>>
>>
>> Your context [from-freeswitch] must have an extension, named as
>> 'freeswitch' (or your DID number) for incoming calls operations.
>>
>>
>>
>>
>> 2016-02-02 21:28 GMT+02:00 Marc S <therebel22 at gmail.com>:
>>
>>> It seems that "freeswitch gateway username" return all results about
>>> setup username and password on FS to register against external SIP gateway,
>>> am i wrong ?
>>>
>>> 2016-02-02 20:23 GMT+01:00 Oz Mortimer <omortimer at gmail.com>:
>>>
>>>> Setup username / password authentication on asterisk and set the
>>>> corresponding user & pass in your freeswitch gateway.
>>>> I'm sure a google for "asterisk username authentication" and
>>>> "freeswitch gateway username" will give you plenty of examples - you will
>>>> want freeswitch to bridge to gateway - again Google "freeswitch bridge
>>>> gateway".
>>>> Google is your friend..
>>>>
>>>>
>>>> On 2 Feb 2016, at 19:03, Marc S <therebel22 at gmail.com> wrote:
>>>>
>>>> I want to use mod xml curl to generate dynamic dialplan xml like this :
>>>>
>>>>  <extension name="inbound_12345678">
>>>>         <condition field="destination_number" expression="^(12345678)$">
>>>>            <action application="bridge" data="user/myuserid"/>
>>>>         </condition>
>>>> </extension>
>>>>
>>>> Thanks
>>>>
>>>>
>>>>
>>>> 2016-02-02 19:16 GMT+01:00 Sergey Safarov <s.safarov at gmail.com>:
>>>>
>>>>> What is way you planing to use for link DID with user?
>>>>>
>>>>> Sergey
>>>>>
>>>>> On Tue, Feb 2, 2016 at 8:20 PM, Marc S <therebel22 at gmail.com> wrote:
>>>>>
>>>>>> Hello,
>>>>>> sorry for asking again :
>>>>>>
>>>>>> I hav an asterisk that register on Freeswitch (as a user).
>>>>>>
>>>>>> When a call is incoming to FS, FS send it to asterisk : In asterisk,
>>>>>> it is the s extension.
>>>>>>
>>>>>> Here my bridge tests
>>>>>>
>>>>>> <action application="bridge" data="user/myuserid"/>
>>>>>>
>>>>>> => s extension in asterisk instead of extension
>>>>>>
>>>>>> <action application="bridge" data="sofia/mycontext/myextension@
>>>>>> <ASTERISK_IP>"/>
>>>>>>
>>>>>> => Not authenticated in asterisk (because no IP authentication in
>>>>>> asterisk)
>>>>>>
>>>>>> Have you an idea how to send real extension instead of s extension  ?
>>>>>> Thanks
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> 2016-01-03 10:45 GMT+01:00 Marc S <therebel22 at gmail.com>:
>>>>>>
>>>>>>> Hello,
>>>>>>>
>>>>>>> i'm discovering FS. I hav read a lot about users and gateways.
>>>>>>>
>>>>>>> I would like to  FS act as registrar for authenticated SIP trunking.
>>>>>>>
>>>>>>> - Customers IPBX would register with login/password to Freeswitch.
>>>>>>> - Incoming call would be routed to these SIP trunks in dialplan XML.
>>>>>>>
>>>>>>> directory/users does not seem to be the solution because in
>>>>>>> dialplan, destination DID can't be defined, only user id :
>>>>>>>
>>>>>>> <action application="bridge" data="user/myuserid"/>
>>>>>>>
>>>>>>> gateway seems to be designed for SIP trunking to remote SIP gateway,
>>>>>>> not for FS to act as registrar.
>>>>>>>
>>>>>>> Is it possible to FS to act as authenticated SIP trunking registrar ?
>>>>>>>
>>>>>>> Thanks a lot,
>>>>>>> Marc
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://
>>>> lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

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 omortimer at gmail.com  Tue Feb  2 23:29:54 2016
From: omortimer at gmail.com (Oz Mortimer)
Date: Tue, 2 Feb 2016 20:29:54 +0000
Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk
In-Reply-To: <CAMs117W+g2cFaQ-N0ULKMS-EDywazOsvfakBaQ4Hzv8VZ4LGnw@mail.gmail.com>
References: <CAMs117X_s+BHbQkPww0jMdC4Jf0z5aV=dy9KAYm3NLc-ojqsww@mail.gmail.com>
	<CAMs117V-c0hSBJrWBxQkgoxndpVKKMtpbYMbUV_Q5RwC+SSXRA@mail.gmail.com>
	<CAHtxdDeKv40QTvRsU6jbewLb-ikOXUVihgpKDCvixiFzuVrUbg@mail.gmail.com>
	<CAMs117X0Y2xwZ6EPMvaPQU9m6tLYswb_8SHte_9bdXDhKa8R8w@mail.gmail.com>
	<490EAD37-3FE1-4A94-8166-499B461F9C53@gmail.com>
	<CAMs117W+g2cFaQ-N0ULKMS-EDywazOsvfakBaQ4Hzv8VZ4LGnw@mail.gmail.com>
Message-ID: <1CB79FBF-A771-4A02-AD43-70597741BA9D@gmail.com>

I think you need to explain what you are trying to achieve.. 
What you are doing currently is registering a user against freeswitch - not dissimilar to logging in to Skype. 
In your bridge statement you are calling the registered user - again like you would in Skype.
Is there a reason you want asterisk to register against freeswitch? The only thing I can think of is that it's on a private LAN and that your asterisk box does something "special"..


> On 2 Feb 2016, at 19:28, Marc S <therebel22 at gmail.com> wrote:
> 
> It seems that "freeswitch gateway username" return all results about setup username and password on FS to register against external SIP gateway, am i wrong ?
> 
> 2016-02-02 20:23 GMT+01:00 Oz Mortimer <omortimer at gmail.com>:
>> Setup username / password authentication on asterisk and set the corresponding user & pass in your freeswitch gateway. 
>> I'm sure a google for "asterisk username authentication" and "freeswitch gateway username" will give you plenty of examples - you will want freeswitch to bridge to gateway - again Google "freeswitch bridge gateway".
>> Google is your friend.. 
>> 
>> 
>>> On 2 Feb 2016, at 19:03, Marc S <therebel22 at gmail.com> wrote:
>>> 
>>> I want to use mod xml curl to generate dynamic dialplan xml like this :
>>> 
>>>  <extension name="inbound_12345678">
>>>         <condition field="destination_number" expression="^(12345678)$">
>>>            <action application="bridge" data="user/myuserid"/>
>>>         </condition>
>>> </extension>
>>> 
>>> Thanks
>>> 
>>> 
>>> 
>>> 2016-02-02 19:16 GMT+01:00 Sergey Safarov <s.safarov at gmail.com>:
>>>> What is way you planing to use for link DID with user?
>>>> 
>>>> Sergey
>>>> 
>>>>> On Tue, Feb 2, 2016 at 8:20 PM, Marc S <therebel22 at gmail.com> wrote:
>>>>> Hello,
>>>>> sorry for asking again :
>>>>> 
>>>>> I hav an asterisk that register on Freeswitch (as a user). 
>>>>> 
>>>>> When a call is incoming to FS, FS send it to asterisk : In asterisk, it is the s extension.
>>>>> 
>>>>> Here my bridge tests
>>>>> 
>>>>> <action application="bridge" data="user/myuserid"/> 
>>>>> 
>>>>> => s extension in asterisk instead of extension
>>>>> 
>>>>> <action application="bridge" data="sofia/mycontext/myextension@<ASTERISK_IP>"/>
>>>>> 
>>>>> => Not authenticated in asterisk (because no IP authentication in asterisk)
>>>>> 
>>>>> Have you an idea how to send real extension instead of s extension  ?
>>>>> Thanks
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> 2016-01-03 10:45 GMT+01:00 Marc S <therebel22 at gmail.com>:
>>>>>> Hello,
>>>>>> 
>>>>>> i'm discovering FS. I hav read a lot about users and gateways.
>>>>>> 
>>>>>> I would like to  FS act as registrar for authenticated SIP trunking.
>>>>>> 
>>>>>> - Customers IPBX would register with login/password to Freeswitch. 
>>>>>> - Incoming call would be routed to these SIP trunks in dialplan XML.
>>>>>> 
>>>>>> directory/users does not seem to be the solution because in dialplan, destination DID can't be defined, only user id :
>>>>>> 
>>>>>> <action application="bridge" data="user/myuserid"/>
>>>>>> 
>>>>>> gateway seems to be designed for SIP trunking to remote SIP gateway, not for FS to act as registrar.
>>>>>> 
>>>>>> Is it possible to FS to act as authenticated SIP trunking registrar ?
>>>>>> 
>>>>>> Thanks a lot,
>>>>>> Marc
>>>>> 
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services: 
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 therebel22 at gmail.com  Tue Feb  2 23:42:09 2016
From: therebel22 at gmail.com (Marc S)
Date: Tue, 2 Feb 2016 21:42:09 +0100
Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk
In-Reply-To: <1CB79FBF-A771-4A02-AD43-70597741BA9D@gmail.com>
References: <CAMs117X_s+BHbQkPww0jMdC4Jf0z5aV=dy9KAYm3NLc-ojqsww@mail.gmail.com>
	<CAMs117V-c0hSBJrWBxQkgoxndpVKKMtpbYMbUV_Q5RwC+SSXRA@mail.gmail.com>
	<CAHtxdDeKv40QTvRsU6jbewLb-ikOXUVihgpKDCvixiFzuVrUbg@mail.gmail.com>
	<CAMs117X0Y2xwZ6EPMvaPQU9m6tLYswb_8SHte_9bdXDhKa8R8w@mail.gmail.com>
	<490EAD37-3FE1-4A94-8166-499B461F9C53@gmail.com>
	<CAMs117W+g2cFaQ-N0ULKMS-EDywazOsvfakBaQ4Hzv8VZ4LGnw@mail.gmail.com>
	<1CB79FBF-A771-4A02-AD43-70597741BA9D@gmail.com>
Message-ID: <CAMs117WrOOSBAXeh12Nq9Zw3f2RuVByxe3VaA-J0Tm3-A7-WSg@mail.gmail.com>

Thanks for all replies.

FS would be used as SBC.

I would like my customers' IPBX (Asterisk, Alcatel, ..)  register against
FS, as a SIP trunk and then i would like routing several DID (not only s)
on incoming call to FS  to customers IPBX with bridge action.

I prefer IPBX login/pass auth against FS rather than IPBX IP auth only,
because of security.

But registring user in FS seems to expect SIP phone only as user to
register, not IPBX as user :

Here my asterisk Register :

REGISTER sip:1.2.3.4 SIP/2.0
Via: SIP/2.0/UDP 5.6.7.8:5060;branch=z9hG4bK45872adf;rport
From: <sip:testipbx2 at 1.2.3.4>;tag=as47a2d14b
To: <sip:testipbx2 at 1.2.3.4>
Call-ID: 6896d4fe6ab4904b7830df881a25e4cf at 127.0.1.1
CSeq: 103 REGISTER
User-Agent: Asterisk
Max-Forwards: 70
Authorization: Digest username="testipbx2", realm="1.2.3.4", algorithm=MD5,
uri="sip:1.2.3.4", nonce="618f279c-c9ea-11e5-9551-4749c6ba93f1",
response="332db701f8c8404f4f1624b9f6c68f2c", qop=auth, cnonce="751ce1f8",
nc=00000001
Expires: 120
Contact: <sip:s at 5.6.7.8>
Event: registration
Content-Length: 0

Here FS->Asterisk invite on incoming call

INVITE sip:s at 5.6.7.8 SIP/2.0
Via: SIP/2.0/UDP 1.2.3.4;rport;branch=z9hG4bKgaUt932FD952c
Max-Forwards: 69
From: "45678776" <sip:45678776 at 1.2.3.4>;tag=4v1S65HQ930mF
To: <sip:s at 5.6.7.8>
Call-ID: 971c11f4-c9ea-11e5-956b-4749c6ba93f1
CSeq: 86886190 INVITE
Contact: <sip:mod_sofia at 1.2.3.4:5060>
User-Agent: FS/1.6
..
Thanks





2016-02-02 21:29 GMT+01:00 Oz Mortimer <omortimer at gmail.com>:

> I think you need to explain what you are trying to achieve..
> What you are doing currently is registering a user against freeswitch -
> not dissimilar to logging in to Skype.
> In your bridge statement you are calling the registered user - again like
> you would in Skype.
> Is there a reason you want asterisk to register against freeswitch? The
> only thing I can think of is that it's on a private LAN and that your
> asterisk box does something "special"..
>
>
> On 2 Feb 2016, at 19:28, Marc S <therebel22 at gmail.com> wrote:
>
> It seems that "freeswitch gateway username" return all results about setup
> username and password on FS to register against external SIP gateway, am i
> wrong ?
>
> 2016-02-02 20:23 GMT+01:00 Oz Mortimer <omortimer at gmail.com>:
>
>> Setup username / password authentication on asterisk and set the
>> corresponding user & pass in your freeswitch gateway.
>> I'm sure a google for "asterisk username authentication" and "freeswitch
>> gateway username" will give you plenty of examples - you will want
>> freeswitch to bridge to gateway - again Google "freeswitch bridge gateway".
>> Google is your friend..
>>
>>
>> On 2 Feb 2016, at 19:03, Marc S <therebel22 at gmail.com> wrote:
>>
>> I want to use mod xml curl to generate dynamic dialplan xml like this :
>>
>>  <extension name="inbound_12345678">
>>         <condition field="destination_number" expression="^(12345678)$">
>>            <action application="bridge" data="user/myuserid"/>
>>         </condition>
>> </extension>
>>
>> Thanks
>>
>>
>>
>> 2016-02-02 19:16 GMT+01:00 Sergey Safarov <s.safarov at gmail.com>:
>>
>>> What is way you planing to use for link DID with user?
>>>
>>> Sergey
>>>
>>> On Tue, Feb 2, 2016 at 8:20 PM, Marc S <therebel22 at gmail.com> wrote:
>>>
>>>> Hello,
>>>> sorry for asking again :
>>>>
>>>> I hav an asterisk that register on Freeswitch (as a user).
>>>>
>>>> When a call is incoming to FS, FS send it to asterisk : In asterisk, it
>>>> is the s extension.
>>>>
>>>> Here my bridge tests
>>>>
>>>> <action application="bridge" data="user/myuserid"/>
>>>>
>>>> => s extension in asterisk instead of extension
>>>>
>>>> <action application="bridge" data="sofia/mycontext/myextension@
>>>> <ASTERISK_IP>"/>
>>>>
>>>> => Not authenticated in asterisk (because no IP authentication in
>>>> asterisk)
>>>>
>>>> Have you an idea how to send real extension instead of s extension  ?
>>>> Thanks
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> 2016-01-03 10:45 GMT+01:00 Marc S <therebel22 at gmail.com>:
>>>>
>>>>> Hello,
>>>>>
>>>>> i'm discovering FS. I hav read a lot about users and gateways.
>>>>>
>>>>> I would like to  FS act as registrar for authenticated SIP trunking.
>>>>>
>>>>> - Customers IPBX would register with login/password to Freeswitch.
>>>>> - Incoming call would be routed to these SIP trunks in dialplan XML.
>>>>>
>>>>> directory/users does not seem to be the solution because in dialplan,
>>>>> destination DID can't be defined, only user id :
>>>>>
>>>>> <action application="bridge" data="user/myuserid"/>
>>>>>
>>>>> gateway seems to be designed for SIP trunking to remote SIP gateway,
>>>>> not for FS to act as registrar.
>>>>>
>>>>> Is it possible to FS to act as authenticated SIP trunking registrar ?
>>>>>
>>>>> Thanks a lot,
>>>>> Marc
>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 jprangi at gmail.com  Tue Feb  2 23:47:50 2016
From: jprangi at gmail.com (Jai Rangi)
Date: Tue, 2 Feb 2016 12:47:50 -0800
Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk
In-Reply-To: <1CB79FBF-A771-4A02-AD43-70597741BA9D@gmail.com>
References: <CAMs117X_s+BHbQkPww0jMdC4Jf0z5aV=dy9KAYm3NLc-ojqsww@mail.gmail.com>
	<CAMs117V-c0hSBJrWBxQkgoxndpVKKMtpbYMbUV_Q5RwC+SSXRA@mail.gmail.com>
	<CAHtxdDeKv40QTvRsU6jbewLb-ikOXUVihgpKDCvixiFzuVrUbg@mail.gmail.com>
	<CAMs117X0Y2xwZ6EPMvaPQU9m6tLYswb_8SHte_9bdXDhKa8R8w@mail.gmail.com>
	<490EAD37-3FE1-4A94-8166-499B461F9C53@gmail.com>
	<CAMs117W+g2cFaQ-N0ULKMS-EDywazOsvfakBaQ4Hzv8VZ4LGnw@mail.gmail.com>
	<1CB79FBF-A771-4A02-AD43-70597741BA9D@gmail.com>
Message-ID: <CAPfjz3hh9XSoBDgzpfkjeo92UyEdy_-vf3OW9dUmDwxLbZJt9A@mail.gmail.com>

We used LUA script to deal with registered asterisk systems.
Example DID 888 555 6666 is assigned to user 101, registered on some ip say
8.8.8.8
Contact will look like
s at 8.8.8.8
or
101 at 8.8.8.8.8

LUA Code
DIALED_NUMBER=8885556666
USER = 111
location = api:execute("sofia_contact", USER) or ""
destinationlocation=location:gsub("sip:(.-)@","sip:"..DIALED_NUMBER.."@")
-- This will replace user part in contact heard, (s or even username) with
the DID number.
-- destinationlocation will look like sip:8885556666 at 8.8.8.8

if (destinationlocation ~= "") then

session:execute("bridge","{sip_cid_type=rpid,loop=3}"..destinationlocation)

end

Hope this help.

*Jai Rangi*
Cebod Technologies LLC dba DIDforSale/Cebod Telecom
O 949-471-0102 <O%20949-471-0101> | C 949-419-7634 <C%20949-742-2666> | F
949-269-0449 / 949-232-1410 | jprangi at didforsale.com  www.cebod.com |
www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA 92626






On Tue, Feb 2, 2016 at 12:29 PM, Oz Mortimer <omortimer at gmail.com> wrote:

> I think you need to explain what you are trying to achieve..
> What you are doing currently is registering a user against freeswitch -
> not dissimilar to logging in to Skype.
> In your bridge statement you are calling the registered user - again like
> you would in Skype.
> Is there a reason you want asterisk to register against freeswitch? The
> only thing I can think of is that it's on a private LAN and that your
> asterisk box does something "special"..
>
>
> On 2 Feb 2016, at 19:28, Marc S <therebel22 at gmail.com> wrote:
>
> It seems that "freeswitch gateway username" return all results about setup
> username and password on FS to register against external SIP gateway, am i
> wrong ?
>
> 2016-02-02 20:23 GMT+01:00 Oz Mortimer <omortimer at gmail.com>:
>
>> Setup username / password authentication on asterisk and set the
>> corresponding user & pass in your freeswitch gateway.
>> I'm sure a google for "asterisk username authentication" and "freeswitch
>> gateway username" will give you plenty of examples - you will want
>> freeswitch to bridge to gateway - again Google "freeswitch bridge gateway".
>> Google is your friend..
>>
>>
>> On 2 Feb 2016, at 19:03, Marc S <therebel22 at gmail.com> wrote:
>>
>> I want to use mod xml curl to generate dynamic dialplan xml like this :
>>
>>  <extension name="inbound_12345678">
>>         <condition field="destination_number" expression="^(12345678)$">
>>            <action application="bridge" data="user/myuserid"/>
>>         </condition>
>> </extension>
>>
>> Thanks
>>
>>
>>
>> 2016-02-02 19:16 GMT+01:00 Sergey Safarov <s.safarov at gmail.com>:
>>
>>> What is way you planing to use for link DID with user?
>>>
>>> Sergey
>>>
>>> On Tue, Feb 2, 2016 at 8:20 PM, Marc S <therebel22 at gmail.com> wrote:
>>>
>>>> Hello,
>>>> sorry for asking again :
>>>>
>>>> I hav an asterisk that register on Freeswitch (as a user).
>>>>
>>>> When a call is incoming to FS, FS send it to asterisk : In asterisk, it
>>>> is the s extension.
>>>>
>>>> Here my bridge tests
>>>>
>>>> <action application="bridge" data="user/myuserid"/>
>>>>
>>>> => s extension in asterisk instead of extension
>>>>
>>>> <action application="bridge" data="sofia/mycontext/myextension@
>>>> <ASTERISK_IP>"/>
>>>>
>>>> => Not authenticated in asterisk (because no IP authentication in
>>>> asterisk)
>>>>
>>>> Have you an idea how to send real extension instead of s extension  ?
>>>> Thanks
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> 2016-01-03 10:45 GMT+01:00 Marc S <therebel22 at gmail.com>:
>>>>
>>>>> Hello,
>>>>>
>>>>> i'm discovering FS. I hav read a lot about users and gateways.
>>>>>
>>>>> I would like to  FS act as registrar for authenticated SIP trunking.
>>>>>
>>>>> - Customers IPBX would register with login/password to Freeswitch.
>>>>> - Incoming call would be routed to these SIP trunks in dialplan XML.
>>>>>
>>>>> directory/users does not seem to be the solution because in dialplan,
>>>>> destination DID can't be defined, only user id :
>>>>>
>>>>> <action application="bridge" data="user/myuserid"/>
>>>>>
>>>>> gateway seems to be designed for SIP trunking to remote SIP gateway,
>>>>> not for FS to act as registrar.
>>>>>
>>>>> Is it possible to FS to act as authenticated SIP trunking registrar ?
>>>>>
>>>>> Thanks a lot,
>>>>> Marc
>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Tue Feb  2 23:50:22 2016
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Feb 2016 14:50:22 -0600
Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk
In-Reply-To: <CAMs117WrOOSBAXeh12Nq9Zw3f2RuVByxe3VaA-J0Tm3-A7-WSg@mail.gmail.com>
References: <CAMs117X_s+BHbQkPww0jMdC4Jf0z5aV=dy9KAYm3NLc-ojqsww@mail.gmail.com>
	<CAMs117V-c0hSBJrWBxQkgoxndpVKKMtpbYMbUV_Q5RwC+SSXRA@mail.gmail.com>
	<CAHtxdDeKv40QTvRsU6jbewLb-ikOXUVihgpKDCvixiFzuVrUbg@mail.gmail.com>
	<CAMs117X0Y2xwZ6EPMvaPQU9m6tLYswb_8SHte_9bdXDhKa8R8w@mail.gmail.com>
	<490EAD37-3FE1-4A94-8166-499B461F9C53@gmail.com>
	<CAMs117W+g2cFaQ-N0ULKMS-EDywazOsvfakBaQ4Hzv8VZ4LGnw@mail.gmail.com>
	<1CB79FBF-A771-4A02-AD43-70597741BA9D@gmail.com>
	<CAMs117WrOOSBAXeh12Nq9Zw3f2RuVByxe3VaA-J0Tm3-A7-WSg@mail.gmail.com>
Message-ID: <CAEJMVkC1BQF9m3E3+U7g4EwQKWdTYqdUL9-wHnUnPh178kPnZA@mail.gmail.com>

You seem like you're trying to fly before you can crawl with FreeSWITCH.
You have some base level technological understanding to acquire prior to
getting to the end results. You probably didn't read what I wrote in my
previous email.   Your device is registering as sip:s at 5.6.7.8, so we're
gonna follow the rules and call you back at sip:s at 5.6.7.8

You can however in FreeSWITCH force the auth user to match the user and
there by having exactly one DID per sip account, or you could change the
RURI (which an example of how to do this is in the vanilla config we ship)

Here it is:

    <!--



        This is an example of how to override the RURI on an outgoing
invite to a registered contact.


    -->




    <extension name="ruri">



      <condition field="destination_number" expression="^ruri$">



        <action application="bridge" data="sofia/${ruri_profile}/${ruri
_user}${regex(${sofia_contact(${ruri_contact})}|^[^\@]+(.*)|%1)}"/>


      </condition>



    </extension>







    <extension name="7004">



      <condition field="destination_number" expression="^7004$">



        <action application="set" data="ruri_profile=default"/>



        <action application="set" data="ruri_user=2000"/>



        <action application="set" data="ruri_contact=1001@${domain_name}"/>



        <action application="execute_extension" data="ruri"/>



      </condition>



    </extension>



On Tue, Feb 2, 2016 at 2:42 PM, Marc S <therebel22 at gmail.com> wrote:

> Thanks for all replies.
>
> FS would be used as SBC.
>
> I would like my customers' IPBX (Asterisk, Alcatel, ..)  register against
> FS, as a SIP trunk and then i would like routing several DID (not only s)
> on incoming call to FS  to customers IPBX with bridge action.
>
> I prefer IPBX login/pass auth against FS rather than IPBX IP auth only,
> because of security.
>
> But registring user in FS seems to expect SIP phone only as user to
> register, not IPBX as user :
>
> Here my asterisk Register :
>
> REGISTER sip:1.2.3.4 SIP/2.0
> Via: SIP/2.0/UDP 5.6.7.8:5060;branch=z9hG4bK45872adf;rport
> From: <sip:testipbx2 at 1.2.3.4>;tag=as47a2d14b
> To: <sip:testipbx2 at 1.2.3.4>
> Call-ID: 6896d4fe6ab4904b7830df881a25e4cf at 127.0.1.1
> CSeq: 103 REGISTER
> User-Agent: Asterisk
> Max-Forwards: 70
> Authorization: Digest username="testipbx2", realm="1.2.3.4",
> algorithm=MD5, uri="sip:1.2.3.4",
> nonce="618f279c-c9ea-11e5-9551-4749c6ba93f1",
> response="332db701f8c8404f4f1624b9f6c68f2c", qop=auth, cnonce="751ce1f8",
> nc=00000001
> Expires: 120
> Contact: <sip:s at 5.6.7.8>
> Event: registration
> Content-Length: 0
>
> Here FS->Asterisk invite on incoming call
>
> INVITE sip:s at 5.6.7.8 SIP/2.0
> Via: SIP/2.0/UDP 1.2.3.4;rport;branch=z9hG4bKgaUt932FD952c
> Max-Forwards: 69
> From: "45678776" <sip:45678776 at 1.2.3.4>;tag=4v1S65HQ930mF
> To: <sip:s at 5.6.7.8>
> Call-ID: 971c11f4-c9ea-11e5-956b-4749c6ba93f1
> CSeq: 86886190 INVITE
> Contact: <sip:mod_sofia at 1.2.3.4:5060>
> User-Agent: FS/1.6
> ..
> Thanks
>
>
>
>
>
> 2016-02-02 21:29 GMT+01:00 Oz Mortimer <omortimer at gmail.com>:
>
>> I think you need to explain what you are trying to achieve..
>> What you are doing currently is registering a user against freeswitch -
>> not dissimilar to logging in to Skype.
>> In your bridge statement you are calling the registered user - again like
>> you would in Skype.
>> Is there a reason you want asterisk to register against freeswitch? The
>> only thing I can think of is that it's on a private LAN and that your
>> asterisk box does something "special"..
>>
>>
>> On 2 Feb 2016, at 19:28, Marc S <therebel22 at gmail.com> wrote:
>>
>> It seems that "freeswitch gateway username" return all results about
>> setup username and password on FS to register against external SIP gateway,
>> am i wrong ?
>>
>> 2016-02-02 20:23 GMT+01:00 Oz Mortimer <omortimer at gmail.com>:
>>
>>> Setup username / password authentication on asterisk and set the
>>> corresponding user & pass in your freeswitch gateway.
>>> I'm sure a google for "asterisk username authentication" and "freeswitch
>>> gateway username" will give you plenty of examples - you will want
>>> freeswitch to bridge to gateway - again Google "freeswitch bridge gateway".
>>> Google is your friend..
>>>
>>>
>>> On 2 Feb 2016, at 19:03, Marc S <therebel22 at gmail.com> wrote:
>>>
>>> I want to use mod xml curl to generate dynamic dialplan xml like this :
>>>
>>>  <extension name="inbound_12345678">
>>>         <condition field="destination_number" expression="^(12345678)$">
>>>            <action application="bridge" data="user/myuserid"/>
>>>         </condition>
>>> </extension>
>>>
>>> Thanks
>>>
>>>
>>>
>>> 2016-02-02 19:16 GMT+01:00 Sergey Safarov <s.safarov at gmail.com>:
>>>
>>>> What is way you planing to use for link DID with user?
>>>>
>>>> Sergey
>>>>
>>>> On Tue, Feb 2, 2016 at 8:20 PM, Marc S <therebel22 at gmail.com> wrote:
>>>>
>>>>> Hello,
>>>>> sorry for asking again :
>>>>>
>>>>> I hav an asterisk that register on Freeswitch (as a user).
>>>>>
>>>>> When a call is incoming to FS, FS send it to asterisk : In asterisk,
>>>>> it is the s extension.
>>>>>
>>>>> Here my bridge tests
>>>>>
>>>>> <action application="bridge" data="user/myuserid"/>
>>>>>
>>>>> => s extension in asterisk instead of extension
>>>>>
>>>>> <action application="bridge" data="sofia/mycontext/myextension@
>>>>> <ASTERISK_IP>"/>
>>>>>
>>>>> => Not authenticated in asterisk (because no IP authentication in
>>>>> asterisk)
>>>>>
>>>>> Have you an idea how to send real extension instead of s extension  ?
>>>>> Thanks
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> 2016-01-03 10:45 GMT+01:00 Marc S <therebel22 at gmail.com>:
>>>>>
>>>>>> Hello,
>>>>>>
>>>>>> i'm discovering FS. I hav read a lot about users and gateways.
>>>>>>
>>>>>> I would like to  FS act as registrar for authenticated SIP trunking.
>>>>>>
>>>>>> - Customers IPBX would register with login/password to Freeswitch.
>>>>>> - Incoming call would be routed to these SIP trunks in dialplan XML.
>>>>>>
>>>>>> directory/users does not seem to be the solution because in dialplan,
>>>>>> destination DID can't be defined, only user id :
>>>>>>
>>>>>> <action application="bridge" data="user/myuserid"/>
>>>>>>
>>>>>> gateway seems to be designed for SIP trunking to remote SIP gateway,
>>>>>> not for FS to act as registrar.
>>>>>>
>>>>>> Is it possible to FS to act as authenticated SIP trunking registrar ?
>>>>>>
>>>>>> Thanks a lot,
>>>>>> Marc
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

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 therebel22 at gmail.com  Wed Feb  3 00:00:10 2016
From: therebel22 at gmail.com (Marc S)
Date: Tue, 2 Feb 2016 22:00:10 +0100
Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk
In-Reply-To: <CAEJMVkC1BQF9m3E3+U7g4EwQKWdTYqdUL9-wHnUnPh178kPnZA@mail.gmail.com>
References: <CAMs117X_s+BHbQkPww0jMdC4Jf0z5aV=dy9KAYm3NLc-ojqsww@mail.gmail.com>
	<CAMs117V-c0hSBJrWBxQkgoxndpVKKMtpbYMbUV_Q5RwC+SSXRA@mail.gmail.com>
	<CAHtxdDeKv40QTvRsU6jbewLb-ikOXUVihgpKDCvixiFzuVrUbg@mail.gmail.com>
	<CAMs117X0Y2xwZ6EPMvaPQU9m6tLYswb_8SHte_9bdXDhKa8R8w@mail.gmail.com>
	<490EAD37-3FE1-4A94-8166-499B461F9C53@gmail.com>
	<CAMs117W+g2cFaQ-N0ULKMS-EDywazOsvfakBaQ4Hzv8VZ4LGnw@mail.gmail.com>
	<1CB79FBF-A771-4A02-AD43-70597741BA9D@gmail.com>
	<CAMs117WrOOSBAXeh12Nq9Zw3f2RuVByxe3VaA-J0Tm3-A7-WSg@mail.gmail.com>
	<CAEJMVkC1BQF9m3E3+U7g4EwQKWdTYqdUL9-wHnUnPh178kPnZA@mail.gmail.com>
Message-ID: <CAMs117UctFELYxNVkOw3H+ZhYE0x=Q3_5r9XHOdOBOamb12BYg@mail.gmail.com>

Thanks a lot for reply ! Sorry for misunderstanding.

I'm going to study what you wrote.

2016-02-02 21:50 GMT+01:00 Brian West <brian at freeswitch.org>:

> You seem like you're trying to fly before you can crawl with FreeSWITCH.
> You have some base level technological understanding to acquire prior to
> getting to the end results. You probably didn't read what I wrote in my
> previous email.   Your device is registering as sip:s at 5.6.7.8, so we're
> gonna follow the rules and call you back at sip:s at 5.6.7.8
>
> You can however in FreeSWITCH force the auth user to match the user and
> there by having exactly one DID per sip account, or you could change the
> RURI (which an example of how to do this is in the vanilla config we ship)
>
> Here it is:
>
>     <!--
>
>
>
>         This is an example of how to override the RURI on an outgoing
> invite to a registered contact.
>
>
>     -->
>
>
>
>
>     <extension name="ruri">
>
>
>
>       <condition field="destination_number" expression="^ruri$">
>
>
>
>         <action application="bridge" data="sofia/${ruri_profile}/${ruri
> _user}${regex(${sofia_contact(${ruri_contact})}|^[^\@]+(.*)|%1)}"/>
>
>
>       </condition>
>
>
>
>     </extension>
>
>
>
>
>
>
>
>     <extension name="7004">
>
>
>
>       <condition field="destination_number" expression="^7004$">
>
>
>
>         <action application="set" data="ruri_profile=default"/>
>
>
>
>         <action application="set" data="ruri_user=2000"/>
>
>
>
>         <action application="set" data="ruri_contact=1001@${domain_name}"/>
>
>
>
>         <action application="execute_extension" data="ruri"/>
>
>
>
>       </condition>
>
>
>
>     </extension>
>
>
>
> On Tue, Feb 2, 2016 at 2:42 PM, Marc S <therebel22 at gmail.com> wrote:
>
>> Thanks for all replies.
>>
>> FS would be used as SBC.
>>
>> I would like my customers' IPBX (Asterisk, Alcatel, ..)  register against
>> FS, as a SIP trunk and then i would like routing several DID (not only s)
>> on incoming call to FS  to customers IPBX with bridge action.
>>
>> I prefer IPBX login/pass auth against FS rather than IPBX IP auth only,
>> because of security.
>>
>> But registring user in FS seems to expect SIP phone only as user to
>> register, not IPBX as user :
>>
>> Here my asterisk Register :
>>
>> REGISTER sip:1.2.3.4 SIP/2.0
>> Via: SIP/2.0/UDP 5.6.7.8:5060;branch=z9hG4bK45872adf;rport
>> From: <sip:testipbx2 at 1.2.3.4>;tag=as47a2d14b
>> To: <sip:testipbx2 at 1.2.3.4>
>> Call-ID: 6896d4fe6ab4904b7830df881a25e4cf at 127.0.1.1
>> CSeq: 103 REGISTER
>> User-Agent: Asterisk
>> Max-Forwards: 70
>> Authorization: Digest username="testipbx2", realm="1.2.3.4",
>> algorithm=MD5, uri="sip:1.2.3.4",
>> nonce="618f279c-c9ea-11e5-9551-4749c6ba93f1",
>> response="332db701f8c8404f4f1624b9f6c68f2c", qop=auth, cnonce="751ce1f8",
>> nc=00000001
>> Expires: 120
>> Contact: <sip:s at 5.6.7.8>
>> Event: registration
>> Content-Length: 0
>>
>> Here FS->Asterisk invite on incoming call
>>
>> INVITE sip:s at 5.6.7.8 SIP/2.0
>> Via: SIP/2.0/UDP 1.2.3.4;rport;branch=z9hG4bKgaUt932FD952c
>> Max-Forwards: 69
>> From: "45678776" <sip:45678776 at 1.2.3.4>;tag=4v1S65HQ930mF
>> To: <sip:s at 5.6.7.8>
>> Call-ID: 971c11f4-c9ea-11e5-956b-4749c6ba93f1
>> CSeq: 86886190 INVITE
>> Contact: <sip:mod_sofia at 1.2.3.4:5060>
>> User-Agent: FS/1.6
>> ..
>> Thanks
>>
>>
>>
>>
>>
>> 2016-02-02 21:29 GMT+01:00 Oz Mortimer <omortimer at gmail.com>:
>>
>>> I think you need to explain what you are trying to achieve..
>>> What you are doing currently is registering a user against freeswitch -
>>> not dissimilar to logging in to Skype.
>>> In your bridge statement you are calling the registered user - again
>>> like you would in Skype.
>>> Is there a reason you want asterisk to register against freeswitch? The
>>> only thing I can think of is that it's on a private LAN and that your
>>> asterisk box does something "special"..
>>>
>>>
>>> On 2 Feb 2016, at 19:28, Marc S <therebel22 at gmail.com> wrote:
>>>
>>> It seems that "freeswitch gateway username" return all results about
>>> setup username and password on FS to register against external SIP gateway,
>>> am i wrong ?
>>>
>>> 2016-02-02 20:23 GMT+01:00 Oz Mortimer <omortimer at gmail.com>:
>>>
>>>> Setup username / password authentication on asterisk and set the
>>>> corresponding user & pass in your freeswitch gateway.
>>>> I'm sure a google for "asterisk username authentication" and
>>>> "freeswitch gateway username" will give you plenty of examples - you will
>>>> want freeswitch to bridge to gateway - again Google "freeswitch bridge
>>>> gateway".
>>>> Google is your friend..
>>>>
>>>>
>>>> On 2 Feb 2016, at 19:03, Marc S <therebel22 at gmail.com> wrote:
>>>>
>>>> I want to use mod xml curl to generate dynamic dialplan xml like this :
>>>>
>>>>  <extension name="inbound_12345678">
>>>>         <condition field="destination_number" expression="^(12345678)$">
>>>>            <action application="bridge" data="user/myuserid"/>
>>>>         </condition>
>>>> </extension>
>>>>
>>>> Thanks
>>>>
>>>>
>>>>
>>>> 2016-02-02 19:16 GMT+01:00 Sergey Safarov <s.safarov at gmail.com>:
>>>>
>>>>> What is way you planing to use for link DID with user?
>>>>>
>>>>> Sergey
>>>>>
>>>>> On Tue, Feb 2, 2016 at 8:20 PM, Marc S <therebel22 at gmail.com> wrote:
>>>>>
>>>>>> Hello,
>>>>>> sorry for asking again :
>>>>>>
>>>>>> I hav an asterisk that register on Freeswitch (as a user).
>>>>>>
>>>>>> When a call is incoming to FS, FS send it to asterisk : In asterisk,
>>>>>> it is the s extension.
>>>>>>
>>>>>> Here my bridge tests
>>>>>>
>>>>>> <action application="bridge" data="user/myuserid"/>
>>>>>>
>>>>>> => s extension in asterisk instead of extension
>>>>>>
>>>>>> <action application="bridge" data="sofia/mycontext/myextension@
>>>>>> <ASTERISK_IP>"/>
>>>>>>
>>>>>> => Not authenticated in asterisk (because no IP authentication in
>>>>>> asterisk)
>>>>>>
>>>>>> Have you an idea how to send real extension instead of s extension  ?
>>>>>> Thanks
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> 2016-01-03 10:45 GMT+01:00 Marc S <therebel22 at gmail.com>:
>>>>>>
>>>>>>> Hello,
>>>>>>>
>>>>>>> i'm discovering FS. I hav read a lot about users and gateways.
>>>>>>>
>>>>>>> I would like to  FS act as registrar for authenticated SIP trunking.
>>>>>>>
>>>>>>> - Customers IPBX would register with login/password to Freeswitch.
>>>>>>> - Incoming call would be routed to these SIP trunks in dialplan XML.
>>>>>>>
>>>>>>> directory/users does not seem to be the solution because in
>>>>>>> dialplan, destination DID can't be defined, only user id :
>>>>>>>
>>>>>>> <action application="bridge" data="user/myuserid"/>
>>>>>>>
>>>>>>> gateway seems to be designed for SIP trunking to remote SIP gateway,
>>>>>>> not for FS to act as registrar.
>>>>>>>
>>>>>>> Is it possible to FS to act as authenticated SIP trunking registrar ?
>>>>>>>
>>>>>>> Thanks a lot,
>>>>>>> Marc
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://
>>>> lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> 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 govoiper at gmail.com  Wed Feb  3 03:44:09 2016
From: govoiper at gmail.com (SamyGo)
Date: Tue, 2 Feb 2016 19:44:09 -0500
Subject: [Freeswitch-users] Storm of NOTIFY from FS
Message-ID: <CAJUJwtjEAu_qHoRhwrfQa=oZN-JcQ4uYLkhBg+8Hjm6xuGhp-Q@mail.gmail.com>

Hi All,
Ive few FS servers which solely handle Parking lots and their states behind
Kamailio SBC. It has all been running very good for about two years now.
Just as of this morning all of those FS servers have been sending storm of
NOTIFY messages to Kamailio servers. This in turn consumes all CPU of FS
server and fs cli becomes just irresponsive.

I'm investigating whether Kamailio have blocked FS servers via pike module
and hence FS servers go crazy on resending NOTIFY ever more aggressively.
So far it doesnt look like it.

I just want to ask under what circumstances that even after restarting
service FS it would pick on all the states and resume sending to SBC. Is
there a cache/DB somehwere which causes new  FS process to continue with
NOTIFY storm ?

Ive FS version 1.5 deployed.

Thanks in advance for any help.

Regards,
Sammy
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From nandy1925 at gmail.com  Wed Feb  3 09:16:57 2016
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Wed, 3 Feb 2016 06:16:57 +0000
Subject: [Freeswitch-users] FreeSWITCH vs Asterisk on Reddit...
In-Reply-To: <CAAkW=FdqD5QHYREkyZ4Gf7CF1BH_AeotbA5O8j+HLusC4NCBiQ@mail.gmail.com>
References: <CAEJMVkA3LbRC3B4aYdraogXCM-ytZmkR3sVG_bwaxp5npSgguA@mail.gmail.com>
	<CAKnNFz_Ece7==iYune3YLPuOu_TZXpGCbKYcrfO+-Btqwej9Jg@mail.gmail.com>
	<51f601d137c9$ae032190$0a0964b0$@freeswitch.org>
	<CAAkW=FdqD5QHYREkyZ4Gf7CF1BH_AeotbA5O8j+HLusC4NCBiQ@mail.gmail.com>
Message-ID: <CAHKqMb5pjsJ-WzQPJemY_ZT-risBnxd7ZmYTW4DRSFpq+ATz9A@mail.gmail.com>

My personal experience.  I rescued a client by replacing his Elastix with
FS because it's dropping calls.

On Wed, Dec 16, 2015 at 8:57 PM, Larry Morley <lmorley at neny.cslimits.net>
wrote:

> Regarding this portion of Ken's comment in particular:
>
> "... people think that before there is Free in the name or that it is open
> source, that the developers of the project are doing this for free. ..."
>
> That attitude by no means applies to open source software alone; it's been
> my experience that you'll most certainly find the same attitudes and
> erroneous beliefs in play, in the realms of free works in general - e.g.,
> free or public domain hardware, software, artwork, music, writing, circuit
> designs. Those attitudes and beliefs are the reason why it's so uncommon
> for companies to offer free phone based technical support anymore. They're
> the reason why when you do find phone based tech support, or, for example,
> report a trouble to a phone company, cable company, ITSP, etc., the
> response you get is tiered. In the former, they know there's a good chance
> that the real reason for the call is because the caller didn't want to
> bother reading the documentation supplied with the product. Which, over
> time, has resulted in less documentation being supplied with many products
> - I've witnessed first hand "the bean counters" opting to not spend what
> they would have in years past to hire technical writers, choosing instead
> to have an intern or recent hire produce something the company can put in
> the box.
>
> I firmly believe these attitudes stem from a sense of entitlement. And a
> belief on the part of some that their time, their problem, their existence,
> is more important than anyone else's.
>
> Fortunately, I also know that history is a cyclical beast, and that at
> some point, the attitudes and fundamental beliefs of perhaps not everyone,
> but at least of the greater society, will likely return, for a while at any
> rate - for history is cyclical - to a point where, bearing in mind that
> only a healthy, whole, grounded person has the ability to give of
> themselves let alone anything with giving - people will tend to be far more
> concerned with what they can contribute to others, both now and for
> posterity - and will choose to act in accordance with their beliefs of
> their own accord, their own free wills - than with what they themselves can
> accumulate and with what others can do for them.
>
> I welcome the arrival of that day.
>
> Larry Morley
>
> On Dec 16, 2015 01:20, "Ken Rice" <krice at freeswitch.org> wrote:
> >
> > That guy got cranky because he thought that the developers time was
> free? For some reason people think that before there is Free in the name or
> that it is open source, that the developers of the project are doing this
> for free. We all have families to feed. Unlike some open source projects,
> FreeSWITCH is not funded by millions of dollars of funding, every bit of
> funding comes from people using FreeSWITCH and contracting the FreeSWITCH
> Team to help them deploy, configure, or enhance the software. I can?t think
> of anyone of the supporters that has wanted to keep any code additions out
> of tree.
> >
> >
> >
> > So even if you don?t have a huge budget you can still help out Anthony,
> Brian, Mike, Me, and William. Contact us, let us help with your project.
> Sure, there may be a charge for that initial consultation, but it also
> allows us to dedicate time with you to engineer a plan to help you reach
> your goals.
> >
> >
> >
> > Want to send a little Christmas or Hanukkah gift to one of the
> Developers? Visit https://freeswitch.org/core-team/, Our wishlists are
> there.
> >
> > Want to but the dev?s dinner or something Similar? There?s a donate
> button right on the website.
> >
> >
> >
> > Want to help in other ways?
> >
> > Join the Docs Team and help document things on Confluence. We still have
> a several hundred pages on the old wiki that need to be clean up, updated
> and migrated to Confluence.
> >
> > Help us sort thru the bugs and test patches and pull requests. Anyone
> can comment on open tickets and add information? Want to be an official bug
> marshall? Email me (krice at freeswitch.org) or brian at freeswitch.org off
> list and we?ll help get you started.
> >
> > Join the FreeSWITCH Team via HipChat at https://hipchat.freeswitch.org/
> hang out meet the devs and chat with other FreeSWITCH users.
> >
> > Like IRC? Check out #FreeSWITCH on freenode.net
> >
> >
> >
> > Every Little Bit Helps!
> >
> >
> >
> > K
> >
> >
> >
> >
> >
> >
> >
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of chris
> > Sent: Tuesday, December 15, 2015 11:38 PM
> > To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> > Subject: Re: [Freeswitch-users] FreeSWITCH vs Asterisk on Reddit...
> >
> >
> >
> > about on par with this one which was interesting:
> >
> >
> >
> >
> https://www.reddit.com/r/freeswitch/comments/3o5acr/i_failed_with_freeswitch/
> >
> >
> >
> > On Tue, Dec 15, 2015 at 6:12 PM, Brian West <brian at freeswitch.org>
> wrote:
> >>
> >> Interesting thread...
> >>
> >>
> >>
> >> https://www.reddit.com/r/VOIP/comments/3wy9h8/freeswitch_vs_asterisk/
> >>
> >>
> >>
> >> Everyone should check it out.
> >>
> >>
> >>
> >> Thanks,
> >>
> >>
> >>
> >>
> >> --
> >>
> >> Brian West
> >> brian at freeswitch.org
> >>
> >> Twitter: @FreeSWITCH , @briankwest
> >> http://www.freeswitchbook.com
> >> http://www.freeswitchcookbook.com
> >>
> >> Got Bugs? Report them here! | Reddit: /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
> >
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Wed Feb  3 09:42:21 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Wed, 3 Feb 2016 09:42:21 +0300
Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk
In-Reply-To: <CAMs117X0Y2xwZ6EPMvaPQU9m6tLYswb_8SHte_9bdXDhKa8R8w@mail.gmail.com>
References: <CAMs117X_s+BHbQkPww0jMdC4Jf0z5aV=dy9KAYm3NLc-ojqsww@mail.gmail.com>
	<CAMs117V-c0hSBJrWBxQkgoxndpVKKMtpbYMbUV_Q5RwC+SSXRA@mail.gmail.com>
	<CAHtxdDeKv40QTvRsU6jbewLb-ikOXUVihgpKDCvixiFzuVrUbg@mail.gmail.com>
	<CAMs117X0Y2xwZ6EPMvaPQU9m6tLYswb_8SHte_9bdXDhKa8R8w@mail.gmail.com>
Message-ID: <CAHtxdDeFZjf1_RvwwOjQdCieZR_20+ha6wQrQXjFjnE1vnDqog@mail.gmail.com>

Read theread "Conectin a SPA3102 as pstn gateway on FS
<http://lists.freeswitch.org/pipermail/freeswitch-users/2015-August/115045.html>
"

{sip_invite_to_uri=<sip:${destination_number}@mydomain.org>}user/
reg_user at mydomain.org

If requred mutal autertication

{sip_auth_username='login_name',sip_auth_password='strong_password_here',sip_invite_to_uri=<sip:${
destination_number}@mydomain.org>}user/ reg_user at mydomain.org


On Tue, Feb 2, 2016 at 10:03 PM, Marc S <therebel22 at gmail.com> wrote:

> I want to use mod xml curl to generate dynamic dialplan xml like this :
>
>  <extension name="inbound_12345678">
>         <condition field="destination_number" expression="^(12345678)$">
>            <action application="bridge" data="user/myuserid"/>
>         </condition>
> </extension>
>
> Thanks
>
>
>
> 2016-02-02 19:16 GMT+01:00 Sergey Safarov <s.safarov at gmail.com>:
>
>> What is way you planing to use for link DID with user?
>>
>> Sergey
>>
>> On Tue, Feb 2, 2016 at 8:20 PM, Marc S <therebel22 at gmail.com> wrote:
>>
>>> Hello,
>>> sorry for asking again :
>>>
>>> I hav an asterisk that register on Freeswitch (as a user).
>>>
>>> When a call is incoming to FS, FS send it to asterisk : In asterisk, it
>>> is the s extension.
>>>
>>> Here my bridge tests
>>>
>>> <action application="bridge" data="user/myuserid"/>
>>>
>>> => s extension in asterisk instead of extension
>>>
>>> <action application="bridge" data="sofia/mycontext/myextension@
>>> <ASTERISK_IP>"/>
>>>
>>> => Not authenticated in asterisk (because no IP authentication in
>>> asterisk)
>>>
>>> Have you an idea how to send real extension instead of s extension  ?
>>> Thanks
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> 2016-01-03 10:45 GMT+01:00 Marc S <therebel22 at gmail.com>:
>>>
>>>> Hello,
>>>>
>>>> i'm discovering FS. I hav read a lot about users and gateways.
>>>>
>>>> I would like to  FS act as registrar for authenticated SIP trunking.
>>>>
>>>> - Customers IPBX would register with login/password to Freeswitch.
>>>> - Incoming call would be routed to these SIP trunks in dialplan XML.
>>>>
>>>> directory/users does not seem to be the solution because in dialplan,
>>>> destination DID can't be defined, only user id :
>>>>
>>>> <action application="bridge" data="user/myuserid"/>
>>>>
>>>> gateway seems to be designed for SIP trunking to remote SIP gateway,
>>>> not for FS to act as registrar.
>>>>
>>>> Is it possible to FS to act as authenticated SIP trunking registrar ?
>>>>
>>>> Thanks a lot,
>>>> Marc
>>>>
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 therebel22 at gmail.com  Wed Feb  3 11:25:18 2016
From: therebel22 at gmail.com (Marc S)
Date: Wed, 3 Feb 2016 09:25:18 +0100
Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk
In-Reply-To: <CAHtxdDeFZjf1_RvwwOjQdCieZR_20+ha6wQrQXjFjnE1vnDqog@mail.gmail.com>
References: <CAMs117X_s+BHbQkPww0jMdC4Jf0z5aV=dy9KAYm3NLc-ojqsww@mail.gmail.com>
	<CAMs117V-c0hSBJrWBxQkgoxndpVKKMtpbYMbUV_Q5RwC+SSXRA@mail.gmail.com>
	<CAHtxdDeKv40QTvRsU6jbewLb-ikOXUVihgpKDCvixiFzuVrUbg@mail.gmail.com>
	<CAMs117X0Y2xwZ6EPMvaPQU9m6tLYswb_8SHte_9bdXDhKa8R8w@mail.gmail.com>
	<CAHtxdDeFZjf1_RvwwOjQdCieZR_20+ha6wQrQXjFjnE1vnDqog@mail.gmail.com>
Message-ID: <CAMs117XWA6Q-Gp4DTUg6hLO+-HTVxn4Gt70y+xFN35V9N8yN2w@mail.gmail.com>

I will try soon, thanks

2016-02-03 7:42 GMT+01:00 Sergey Safarov <s.safarov at gmail.com>:

> Read theread "Conectin a SPA3102 as pstn gateway on FS
> <http://lists.freeswitch.org/pipermail/freeswitch-users/2015-August/115045.html>
> "
>
> {sip_invite_to_uri=<sip:${destination_number}@mydomain.org>}user/
> reg_user at mydomain.org
>
> If requred mutal autertication
>
>
> {sip_auth_username='login_name',sip_auth_password='strong_password_here',sip_invite_to_uri=<sip:${
> destination_number}@mydomain.org>}user/ reg_user at mydomain.org
>
>
> On Tue, Feb 2, 2016 at 10:03 PM, Marc S <therebel22 at gmail.com> wrote:
>
>> I want to use mod xml curl to generate dynamic dialplan xml like this :
>>
>>  <extension name="inbound_12345678">
>>         <condition field="destination_number" expression="^(12345678)$">
>>            <action application="bridge" data="user/myuserid"/>
>>         </condition>
>> </extension>
>>
>> Thanks
>>
>>
>>
>> 2016-02-02 19:16 GMT+01:00 Sergey Safarov <s.safarov at gmail.com>:
>>
>>> What is way you planing to use for link DID with user?
>>>
>>> Sergey
>>>
>>> On Tue, Feb 2, 2016 at 8:20 PM, Marc S <therebel22 at gmail.com> wrote:
>>>
>>>> Hello,
>>>> sorry for asking again :
>>>>
>>>> I hav an asterisk that register on Freeswitch (as a user).
>>>>
>>>> When a call is incoming to FS, FS send it to asterisk : In asterisk, it
>>>> is the s extension.
>>>>
>>>> Here my bridge tests
>>>>
>>>> <action application="bridge" data="user/myuserid"/>
>>>>
>>>> => s extension in asterisk instead of extension
>>>>
>>>> <action application="bridge" data="sofia/mycontext/myextension@
>>>> <ASTERISK_IP>"/>
>>>>
>>>> => Not authenticated in asterisk (because no IP authentication in
>>>> asterisk)
>>>>
>>>> Have you an idea how to send real extension instead of s extension  ?
>>>> Thanks
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> 2016-01-03 10:45 GMT+01:00 Marc S <therebel22 at gmail.com>:
>>>>
>>>>> Hello,
>>>>>
>>>>> i'm discovering FS. I hav read a lot about users and gateways.
>>>>>
>>>>> I would like to  FS act as registrar for authenticated SIP trunking.
>>>>>
>>>>> - Customers IPBX would register with login/password to Freeswitch.
>>>>> - Incoming call would be routed to these SIP trunks in dialplan XML.
>>>>>
>>>>> directory/users does not seem to be the solution because in dialplan,
>>>>> destination DID can't be defined, only user id :
>>>>>
>>>>> <action application="bridge" data="user/myuserid"/>
>>>>>
>>>>> gateway seems to be designed for SIP trunking to remote SIP gateway,
>>>>> not for FS to act as registrar.
>>>>>
>>>>> Is it possible to FS to act as authenticated SIP trunking registrar ?
>>>>>
>>>>> Thanks a lot,
>>>>> Marc
>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 yadenis at seznam.cz  Wed Feb  3 11:20:11 2016
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Wed, 3 Feb 2016 09:20:11 +0100
Subject: [Freeswitch-users] Verto vs. SIP.js
In-Reply-To: <CAO6UKa2Tm5j2hF5fCnmNpgxT=7G9mpx5jagw_7PyYWAKTXQAGw@mail.gmail.com>
References: <56AF67C9.2010700@level5.de> <95599933.20160201152539@seznam.cz> 
	<003a01d15cfe$d26c3350$774499f0$@botecomm.com>
	<1105687706.20160201163013@seznam.cz>
	<505001d15d0b$623021c0$26906540$@freeswitch.org>
	<CAO6UKa2Tm5j2hF5fCnmNpgxT=7G9mpx5jagw_7PyYWAKTXQAGw@mail.gmail.com>
Message-ID: <628620366.20160203092011@seznam.cz>

Dobr? den,

The fact that Verto normally does not work on me Firefox does joy for me. For example, I have all the stops on "Refresh Media Devices" (the picture in the attachment)
My clients use different browsers and I can not force them to use something specific. 
With sip.js I have no such problems. It works perfectly, wherever there is support webrtc without problems.


-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

pond?l? 1. ?nora 2016, 17:22:24, napsal jste:


With Verto you can also be more creative, see for example: https://freeswitch.org/verto-not-just-for-call-signaling/
It also provides an interesting feature: "attach". e.g. if a tab is closed by mistake, FS can resume automatically the session upon reconnection (with the same ID).
Also being a simple JSON-based protocol you can have people working on the client side even with limited knowledge of SIP.

On 1 February 2016 at 17:12, Ken Rice <krice at freeswitch.org> wrote:
Really? That?s why the whole protocol is OpenSource? not much vendor lock in there? it can be used anywhere? 
 
 
 
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Denis Jakovlev
Sent: Monday, February 1, 2016 9:30 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Verto vs. SIP.js
 
Dobr? den,

i think this Verto its too much VendorLock :) 

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

pond?l? 1. ?nora 2016, 15:42:48, napsal jste:

Denis, what difficulties did you experience with Verto?

Thanks.

Bote


From: Denis Jakovlev
Sent: Monday, 01 February, 2016 09:26
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Verto vs. SIP.js

Dobr? den,

I use sip.js. It turned out to be a lot easier that Verto.

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

pond?l? 1. ?nora 2016, 15:12:25, napsal jste:

> Hi,

> I want to setup a Click-2-Call-Button for our website. Is there any 
> significant difference between Verto-Mod and libraries such as SIP.js?

> I do not really understand the advantage of Verto (if there is any).

> Thanks in advance,

> Thorsten

> _________________________________________________________________________
 

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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From krice at freeswitch.org  Wed Feb  3 13:03:29 2016
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 3 Feb 2016 04:03:29 -0600
Subject: [Freeswitch-users] Verto vs. SIP.js
In-Reply-To: <628620366.20160203092011@seznam.cz>
References: <56AF67C9.2010700@level5.de> <95599933.20160201152539@seznam.cz>
	<003a01d15cfe$d26c3350$774499f0$@botecomm.com>	<1105687706.20160201163013@seznam.cz>	<505001d15d0b$623021c0$26906540$@freeswitch.org>	<CAO6UKa2Tm5j2hF5fCnmNpgxT=7G9mpx5jagw_7PyYWAKTXQAGw@mail.gmail.com>
	<628620366.20160203092011@seznam.cz>
Message-ID: <590d01d15e6a$22651810$672f4830$@freeswitch.org>

Oh So where is the Jira on this it doesn?t work in firefox with the debugging information?

 

If you know of an issue like this you should report it to jira (https://freeswitch.org/jira) so a dev can try to replicate and fix it?

 

You cant expect bugs to get fixed if you aren?t reporting them properly?

 

Go troll somewhere else?.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Denis Jakovlev
Sent: Wednesday, February 3, 2016 2:20 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Verto vs. SIP.js

 

Dobr? den,

The fact that Verto normally does not work on me Firefox does joy for me. For example, I have all the stops on "Refresh Media Devices" (the picture in the attachment)
My clients use different browsers and I can not force them to use something specific. 
With sip.js I have no such problems. It works perfectly, wherever there is support webrtc without problems.


-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

pond?l? 1. ?nora 2016, 17:22:24, napsal jste:

	
With Verto you can also be more creative, see for example:  <https://freeswitch.org/verto-not-just-for-call-signaling/> https://freeswitch.org/verto-not-just-for-call-signaling/
It also provides an interesting feature: "attach". e.g. if a tab is closed by mistake, FS can resume automatically the session upon reconnection (with the same ID).
Also being a simple JSON-based protocol you can have people working on the client side even with limited knowledge of SIP.

On 1 February 2016 at 17:12, Ken Rice < <mailto:krice at freeswitch.org> krice at freeswitch.org> wrote:
Really? That?s why the whole protocol is OpenSource? not much vendor lock in there? it can be used anywhere? 



From:  <mailto:freeswitch-users-bounces at lists.freeswitch.org> freeswitch-users-bounces at lists.freeswitch.org [mailto: <mailto:freeswitch-users-bounces at lists.freeswitch.org> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Denis Jakovlev
Sent: Monday, February 1, 2016 9:30 AM
To: FreeSWITCH Users Help < <mailto:freeswitch-users at lists.freeswitch.org> freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Verto vs. SIP.js

Dobr? den,

i think this Verto its too much VendorLock :) 

-- 
S pozdravem,
Ing.Denis Jakovlev                           
 <http://mob.tel> mob.tel. 775-415-382

pond?l? 1. ?nora 2016, 15:42:48, napsal jste:

	
Denis, what difficulties did you experience with Verto?

Thanks.

Bote


From: Denis Jakovlev
Sent: Monday, 01 February, 2016 09:26
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Verto vs. SIP.js

Dobr? den,

I use sip.js. It turned out to be a lot easier that Verto.

-- 
S pozdravem,
Ing.Denis Jakovlev                           
 <http://mob.tel> mob.tel. 775-415-382

pond?l? 1. ?nora 2016, 15:12:25, napsal jste:

> Hi,

> I want to setup a Click-2-Call-Button for our website. Is there any 
> significant difference between Verto-Mod and libraries such as SIP.js?

> I do not really understand the advantage of Verto (if there is any).

> Thanks in advance,

> Thorsten

> _________________________________________________________________________



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

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

FreeSWITCH-users mailing list
 <mailto:freeswitch-users at lists.freeswitch.org> FreeSWITCH-users at lists.freeswitch.org
 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE: <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://lists.freeswitch.org/mailman/options/freeswitch-users
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From mike at jerris.com  Wed Feb  3 19:18:53 2016
From: mike at jerris.com (Michael Jerris)
Date: Wed, 3 Feb 2016 11:18:53 -0500
Subject: [Freeswitch-users] mod_potaudio eror on opening device
In-Reply-To: <CAPxXe8UYxwqDYo0Z43EwEiHeQ8z0c1Ei5d7P-bXBUYwthJRx2w@mail.gmail.com>
References: <CAPxXe8UrpMDf2bQ48Z5=qXwX+i-fb5ugo6bzAL-CV5yXMkaFug@mail.gmail.com>
	<CAEwTsAjMbwddo+_wEaiEUxnDMgC5iTR1ZRS0nQ8nD8S3JO23FA@mail.gmail.com>
	<CAPxXe8UYxwqDYo0Z43EwEiHeQ8z0c1Ei5d7P-bXBUYwthJRx2w@mail.gmail.com>
Message-ID: <3B5F650F-5704-40CC-AA6D-5414AF05BBE4@jerris.com>

I know that it is available in the source code, I do not know if packages install these or not, I suspect not.

> On Jan 30, 2016, at 6:56 AM, Pete Kay <petedao at gmail.com> wrote:
> 
> Hi Michael
> 
> Thanks alot for your reply.
> 
> portaudio was installed as part of freeswitch.  
> 
> I search and could not find the test program.  May I know where I can find this test program to test?
> 
> P
> 
> On Mon, Jan 25, 2016 at 4:05 AM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> if you build port audio manually do the sample programs work?
> 
> 
> On Sunday, January 24, 2016, Pete Kay <petedao at gmail.com <mailto:petedao at gmail.com>> wrote:
> Hi 
>  <>
> 
> I am not able to get      pa play to work.      could someone please kindly help me out?  I am getting the following ERROR messages: 
> 
> 
>  <>
> freeswitch at vps57327.vps.ovh.ca <>> pa play   /tmp/running.wav
> 
> Expression 'SetApproximateSampleRate( pcm, hwParams, sr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1293
> 
> Expression 'PaAlsaStreamComponent_InitialConfigure( &self->capture, inParams, self->primeBuffers, hwParamsCapture, &realSr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1869
> 
> Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1993
> 
> 2016-01-24 07:02:32.591002 [ERR] mod_portaudio.c:2428 Error opening audio device retrying
> 
> Expression 'SetApproximateSampleRate( pcm, hwParams, sr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1293
> 
> Expression 'PaAlsaStreamComponent_InitialConfigure( &self->capture, inParams, self->primeBuffers, hwParamsCapture, &realSr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1869
> 
> Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1993
> 
> 
> 
> Failed to engage audio device
> 
> 
> 
> 2016-01-24 07:02:34.090957 [ERR] mod_portaudio.c:2435 Can't open audio device
> 
> freeswitch at vps57327.vps.ovh.ca <>> pa outdev #4 
> 
> 
> 
> outdev set to 4
> 
> 
> 
> freeswitch at vps57327.vps.ovh.ca <>> pa indev #5
> 
> 
> 
> indev set to 5
> 
> 
> 
> freeswitch at vps57327.vps.ovh.ca <>> pa play   /tmp/running.wav
> 
> Expression 'SetApproximateSampleRate( pcm, hwParams, sr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1293
> 
> Expression 'PaAlsaStreamComponent_InitialConfigure( &self->playback, outParams, self->primeBuffers, hwParamsPlayback, &realSr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1872
> 
> Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1993
> 
> 2016-01-24 07:02:51.991013 [ERR] mod_portaudio.c:2428 Error opening audio device retrying
> 
> Expression 'SetApproximateSampleRate( pcm, hwParams, sr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1293
> 
> Expression 'PaAlsaStreamComponent_InitialConfigure( &self->playback, outParams, self->primeBuffers, hwParamsPlayback, &realSr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1872
> 
> Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1993
> 
> 
> 
> Failed to engage audio device
> 
> 
> 
> 2016-01-24 07:02:53.491018 [ERR] mod_portaudio.c:2435 Can't open audio device
> 
> freeswitch at vps57327.vps.ovh.ca <>> 
> 
> 
> _________________________________________________________________________
> 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 dcolombo at voismart.it  Wed Feb  3 19:39:51 2016
From: dcolombo at voismart.it (Davide Colombo)
Date: Wed, 3 Feb 2016 17:39:51 +0100 (CET)
Subject: [Freeswitch-users] Verto vs. SIP.js
In-Reply-To: <590d01d15e6a$22651810$672f4830$@freeswitch.org>
References: <56AF67C9.2010700@level5.de> <95599933.20160201152539@seznam.cz>
	<003a01d15cfe$d26c3350$774499f0$@botecomm.com>
	<1105687706.20160201163013@seznam.cz>
	<505001d15d0b$623021c0$26906540$@freeswitch.org>
	<CAO6UKa2Tm5j2hF5fCnmNpgxT=7G9mpx5jagw_7PyYWAKTXQAGw@mail.gmail.com>
	<628620366.20160203092011@seznam.cz>
	<590d01d15e6a$22651810$672f4830$@freeswitch.org>
Message-ID: <1047097771.103803.1454517591330.JavaMail.zimbra@voismart.it>

I reported this bug to jira:

https://freeswitch.org/jira/browse/FS-8805



----- Messaggio originale -----
Da: "Ken Rice" <krice at freeswitch.org>
A: "freeswitch-users" <freeswitch-users at lists.freeswitch.org>
Inviato: Mercoled?, 3 febbraio 2016 11:03:29
Oggetto: Re: [Freeswitch-users] Verto vs. SIP.js

Re: [Freeswitch-users] Verto vs. SIP.js 


Oh So where is the Jira on this it doesn?t work in firefox with the debugging information? 



If you know of an issue like this you should report it to jira ( https://freeswitch.org/jira ) so a dev can try to replicate and fix it? 



You cant expect bugs to get fixed if you aren?t reporting them properly? 



Go troll somewhere else?. 






From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Denis Jakovlev 
Sent: Wednesday, February 3, 2016 2:20 AM 
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Subject: Re: [Freeswitch-users] Verto vs. SIP.js 




Dobr? den, 

The fact that Verto normally does not work on me Firefox does joy for me. For example, I have all the stops on "Refresh Media Devices" (the picture in the attachment) 
My clients use different browsers and I can not force them to use something specific. 
With sip.js I have no such problems. It works perfectly, wherever there is support webrtc without problems. 


-- 
S pozdravem, 
Ing.Denis Jakovlev 
mob.tel. 775-415-382 

pond?l? 1. ?nora 2016, 17:22:24, napsal jste: 	

With Verto you can also be more creative, see for example: https://freeswitch.org/verto-not-just-for-call-signaling/ 
It also provides an interesting feature: "attach". e.g. if a tab is closed by mistake, FS can resume automatically the session upon reconnection (with the same ID). 
Also being a simple JSON-based protocol you can have people working on the client side even with limited knowledge of SIP. 

On 1 February 2016 at 17:12, Ken Rice < krice at freeswitch.org > wrote: 
Really? That?s why the whole protocol is OpenSource? not much vendor lock in there? it can be used anywhere? 



From: freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Denis Jakovlev 
Sent: Monday, February 1, 2016 9:30 AM 
To: FreeSWITCH Users Help < freeswitch-users at lists.freeswitch.org > 
Subject: Re: [Freeswitch-users] Verto vs. SIP.js 

Dobr? den, 

i think this Verto its too much VendorLock :) 

-- 
S pozdravem, 
Ing.Denis Jakovlev 
mob.tel . 775-415-382 

pond?l? 1. ?nora 2016, 15:42:48, napsal jste: 	

Denis, what difficulties did you experience with Verto? 

Thanks. 

Bote 


From: Denis Jakovlev 
Sent: Monday, 01 February, 2016 09:26 
To: FreeSWITCH Users Help 
Subject: Re: [Freeswitch-users] Verto vs. SIP.js 

Dobr? den, 

I use sip.js. It turned out to be a lot easier that Verto. 

-- 
S pozdravem, 
Ing.Denis Jakovlev 
mob.tel . 775-415-382 

pond?l? 1. ?nora 2016, 15:12:25, napsal jste: 

> Hi, 

> I want to setup a Click-2-Call-Button for our website. Is there any 
> significant difference between Verto-Mod and libraries such as SIP.js? 

> I do not really understand the advantage of Verto (if there is any). 

> Thanks in advance, 

> Thorsten 

> _________________________________________________________________________ 



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

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

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



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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at 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  Wed Feb  3 19:45:21 2016
From: mike at jerris.com (Michael Jerris)
Date: Wed, 3 Feb 2016 11:45:21 -0500
Subject: [Freeswitch-users] BLF for Gateway
In-Reply-To: <BD20678C-416F-413D-AC6E-077C888C3745@dissauer.net>
References: <BD20678C-416F-413D-AC6E-077C888C3745@dissauer.net>
Message-ID: <B96DFBB9-05C1-4A14-9297-B8D2D80ABD9B@jerris.com>

You might be able to do something here with display updates.

> On Feb 2, 2016, at 6:58 AM, Roman Dissauer <roman at dissauer.net> wrote:
> 
> Hi All,
> 
> is there a way to get gateway usage in freeswitch on my phones blf?
> I have multiple gateways registered and want to see which one is taken for a particular outbound call.
> 
> Best Regards,
> Roman



From vfclists at gmail.com  Wed Feb  3 20:11:10 2016
From: vfclists at gmail.com (vfclists .)
Date: Wed, 3 Feb 2016 17:11:10 +0000
Subject: [Freeswitch-users] How do you access caller profile fields in the
	XML dialplan?
Message-ID: <CAE3mkNAQQYTh1wULp5ZeibMDCchz4ZFgn0caeWW-mHvHNhe8vw@mail.gmail.com>

What is the syntax for accessing caller profile fields in the dialplan, for
instance if you want to set a channel variable to a caller profile field?

Is there also a way of performing some string manipulation and extraction
on the variables in the XML dialplan?

This is what I am trying to achieve. The CDR from a service provider
contains the CLI from a gateway on the customers premises, but it doesn't
show which extension on the customer's premises the call came from. What I
need is to be able to obtain the extension of the sip device and combine
that with the customer gateway's CLI so that it shows in the CDR record.

eg if the gateway's CLI is 2340 and the extension of the caller is 1001,
corresponding to the UserId on a sipura, I want the CLI passed to the
service provider to be 23401001. When I check the XML cdr in Freeswitch I
see an XML value which in XPath would be accessed as
/callflow/caller_profile/caller_id_number. All the information is therefore
passed onto Freeswitch for use in the call, and what I need is to be able
to access it and change the effective_caller_id_number before bridging the
call.

-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From italo at freeswitch.org  Wed Feb  3 20:34:58 2016
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Wed, 3 Feb 2016 14:34:58 -0300
Subject: [Freeswitch-users] How do you access caller profile fields in
 the XML dialplan?
In-Reply-To: <CAE3mkNAQQYTh1wULp5ZeibMDCchz4ZFgn0caeWW-mHvHNhe8vw@mail.gmail.com>
References: <CAE3mkNAQQYTh1wULp5ZeibMDCchz4ZFgn0caeWW-mHvHNhe8vw@mail.gmail.com>
Message-ID: <CABMX_w9xB1nwtr=bcv+POkHrDwREe0ZxrM8huDOdzVLHewhKxQ@mail.gmail.com>

https://freeswitch.org/confluence/display/FREESWITCH/XML+Dialplan#XMLDialplan-CallerProfileFieldsvs.ChannelVariables

On Wed, Feb 3, 2016 at 2:11 PM, vfclists . <vfclists at gmail.com> wrote:

> What is the syntax for accessing caller profile fields in the dialplan,
> for instance if you want to set a channel variable to a caller profile
> field?
>
> Is there also a way of performing some string manipulation and extraction
> on the variables in the XML dialplan?
>
> This is what I am trying to achieve. The CDR from a service provider
> contains the CLI from a gateway on the customers premises, but it doesn't
> show which extension on the customer's premises the call came from. What I
> need is to be able to obtain the extension of the sip device and combine
> that with the customer gateway's CLI so that it shows in the CDR record.
>
> eg if the gateway's CLI is 2340 and the extension of the caller is 1001,
> corresponding to the UserId on a sipura, I want the CLI passed to the
> service provider to be 23401001. When I check the XML cdr in Freeswitch I
> see an XML value which in XPath would be accessed as
> /callflow/caller_profile/caller_id_number. All the information is therefore
> passed onto Freeswitch for use in the call, and what I need is to be able
> to access it and change the effective_caller_id_number before bridging the
> call.
>
> --
> 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
>



-- 
?talo Rossi
italo at freeswitch.org
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From vfclists at gmail.com  Wed Feb  3 21:15:44 2016
From: vfclists at gmail.com (vfclists .)
Date: Wed, 3 Feb 2016 18:15:44 +0000
Subject: [Freeswitch-users] How do you access caller profile fields in
 the XML dialplan?
In-Reply-To: <CABMX_w9xB1nwtr=bcv+POkHrDwREe0ZxrM8huDOdzVLHewhKxQ@mail.gmail.com>
References: <CAE3mkNAQQYTh1wULp5ZeibMDCchz4ZFgn0caeWW-mHvHNhe8vw@mail.gmail.com>
	<CABMX_w9xB1nwtr=bcv+POkHrDwREe0ZxrM8huDOdzVLHewhKxQ@mail.gmail.com>
Message-ID: <CAE3mkNAXguEMHN10F4cAT0Gs4RmE9aHy0qBhrVzPzOOO4c4BBA@mail.gmail.com>

There are quite a number of variables there, but I can't seem to find tthe
one I want because the actual extension doesn't seem to be used in the
configuration files.

eg. When I check the logs I see output like this  sofia.c:1192
sofia/internal/2048 at 201.182.29.148 Update Caller ID to "Desk 12" <1012>

What I want to do is to combine the 2048 with the 1012 to create the
effective_caller_id_number 20481012. The information is present somewhere
in Freeswitch. The extensions connect to Asterisk because most of the
applications were written for Asterisk, but Asterisk proved to be terrible
behind NAT so a Freeswitch system was added as an intermediate gateway, so
none of the extensions register to the Freeswitch, only the Asterisk.


On 3 February 2016 at 17:34, ?talo Rossi <italo at freeswitch.org> wrote:

>
> https://freeswitch.org/confluence/display/FREESWITCH/XML+Dialplan#XMLDialplan-CallerProfileFieldsvs.ChannelVariables
>
> On Wed, Feb 3, 2016 at 2:11 PM, vfclists . <vfclists at gmail.com> wrote:
>
>> What is the syntax for accessing caller profile fields in the dialplan,
>> for instance if you want to set a channel variable to a caller profile
>> field?
>>
>> Is there also a way of performing some string manipulation and extraction
>> on the variables in the XML dialplan?
>>
>> This is what I am trying to achieve. The CDR from a service provider
>> contains the CLI from a gateway on the customers premises, but it doesn't
>> show which extension on the customer's premises the call came from. What I
>> need is to be able to obtain the extension of the sip device and combine
>> that with the customer gateway's CLI so that it shows in the CDR record.
>>
>> eg if the gateway's CLI is 2340 and the extension of the caller is 1001,
>> corresponding to the UserId on a sipura, I want the CLI passed to the
>> service provider to be 23401001. When I check the XML cdr in Freeswitch I
>> see an XML value which in XPath would be accessed as
>> /callflow/caller_profile/caller_id_number. All the information is therefore
>> passed onto Freeswitch for use in the call, and what I need is to be able
>> to access it and change the effective_caller_id_number before bridging the
>> call.
>>
>> --
>> 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
>>
>
>
>
> --
> ?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
>



-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From bobjectsfreeswitch at gmail.com  Wed Feb  3 22:28:42 2016
From: bobjectsfreeswitch at gmail.com (Bob Hartwig)
Date: Wed, 3 Feb 2016 13:28:42 -0600
Subject: [Freeswitch-users] Best practices for determining if originated
 call is answered by human or voicemail system?
Message-ID: <CAAemr3yoTC5obmo_SoGcyO+AehMOedRyQHSK41_d02dZrq_teA@mail.gmail.com>

I have a client that needs to reliably detect if their outbound calls are
answered by a human or voicemail system, so that they can take different
actions based on that determination.

I looked at the AVMD module documentation at
https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd , and it
seems to indicate that this simply detects a beep, i.e. it does not use
talking / silence heuristics into account to determine if the call is
answered by a human or machine.

Am I correct about that?  How is this intended to be used, do I assume that
the call is answered by a human until this module sends a avmd%3A%3Abeep
event to me?  How do others use this module or other techniques to
determine human / machine answer to outbound calls with Freeswitch?

Thanks!
Bob
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From anthony.minessale at gmail.com  Wed Feb  3 22:45:39 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 3 Feb 2016 13:45:39 -0600
Subject: [Freeswitch-users] Best practices for determining if originated
 call is answered by human or voicemail system?
In-Reply-To: <CAAemr3yoTC5obmo_SoGcyO+AehMOedRyQHSK41_d02dZrq_teA@mail.gmail.com>
References: <CAAemr3yoTC5obmo_SoGcyO+AehMOedRyQHSK41_d02dZrq_teA@mail.gmail.com>
Message-ID: <CAKbxfG93cVZRY51qeHjCU15d=_4YMtxVOQD0t9ZtzgMv=3hH4w@mail.gmail.com>

There is actually a commercial module for AMD offered by FreeSWITCH
Solutions

https://freeswitch.com/cart.php?gid=2


On Wed, Feb 3, 2016 at 1:28 PM, Bob Hartwig <bobjectsfreeswitch at gmail.com>
wrote:

> I have a client that needs to reliably detect if their outbound calls are
> answered by a human or voicemail system, so that they can take different
> actions based on that determination.
>
> I looked at the AVMD module documentation at
> https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd , and it
> seems to indicate that this simply detects a beep, i.e. it does not use
> talking / silence heuristics into account to determine if the call is
> answered by a human or machine.
>
> Am I correct about that?  How is this intended to be used, do I assume
> that the call is answered by a human until this module sends
> a avmd%3A%3Abeep event to me?  How do others use this module or other
> techniques to determine human / machine answer to outbound calls with
> Freeswitch?
>
> Thanks!
> Bob
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 lists at telefaks.de  Wed Feb  3 22:46:38 2016
From: lists at telefaks.de (Peter Steinbach)
Date: Wed, 03 Feb 2016 20:46:38 +0100
Subject: [Freeswitch-users] Best practices for determining if originated
 call is answered by human or voicemail system?
In-Reply-To: <CAAemr3yoTC5obmo_SoGcyO+AehMOedRyQHSK41_d02dZrq_teA@mail.gmail.com>
References: <CAAemr3yoTC5obmo_SoGcyO+AehMOedRyQHSK41_d02dZrq_teA@mail.gmail.com>
Message-ID: <56B2591E.1030404@telefaks.de>

I have had pretty good results with Sangoma Lyra AMD.

Peter


On 02/03/16 20:28, Bob Hartwig wrote:
> I have a client that needs to reliably detect if their outbound calls
> are answered by a human or voicemail system, so that they can take
> different actions based on that determination.
>
> I looked at the AVMD module documentation
> at https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd , and
> it seems to indicate that this simply detects a beep, i.e. it does not
> use talking / silence heuristics into account to determine if the call
> is answered by a human or machine.  
>
> Am I correct about that?  How is this intended to be used, do I assume
> that the call is answered by a human until this module sends
> a avmd%3A%3Abeep event to me?  How do others use this module or other
> techniques to determine human / machine answer to outbound calls with
> Freeswitch? 
>
> Thanks!
> Bob
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de

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From cmrienzo at gmail.com  Wed Feb  3 22:50:58 2016
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Wed, 3 Feb 2016 14:50:58 -0500
Subject: [Freeswitch-users] Best practices for determining if originated
 call is answered by human or voicemail system?
In-Reply-To: <CAAemr3yoTC5obmo_SoGcyO+AehMOedRyQHSK41_d02dZrq_teA@mail.gmail.com>
References: <CAAemr3yoTC5obmo_SoGcyO+AehMOedRyQHSK41_d02dZrq_teA@mail.gmail.com>
Message-ID: <CAD-qRkjhTHdWBxVgm5baWSV+fLFuPyvzYW4JLf+WB_REUVRN5g@mail.gmail.com>

I wouldn't use beep detection in a dialer application, but it could be
useful in something like follow me to reduce the occurrence of voicemails
being left on subscriber phones.

On Wed, Feb 3, 2016 at 2:28 PM, Bob Hartwig <bobjectsfreeswitch at gmail.com>
wrote:

> I have a client that needs to reliably detect if their outbound calls are
> answered by a human or voicemail system, so that they can take different
> actions based on that determination.
>
> I looked at the AVMD module documentation at
> https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd , and it
> seems to indicate that this simply detects a beep, i.e. it does not use
> talking / silence heuristics into account to determine if the call is
> answered by a human or machine.
>
> Am I correct about that?  How is this intended to be used, do I assume
> that the call is answered by a human until this module sends
> a avmd%3A%3Abeep event to me?  How do others use this module or other
> techniques to determine human / machine answer to outbound calls with
> Freeswitch?
>
> Thanks!
> Bob
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Wed Feb  3 22:57:44 2016
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Wed, 3 Feb 2016 20:57:44 +0100
Subject: [Freeswitch-users] How do you access caller profile fields in
 the XML dialplan?
In-Reply-To: <CAE3mkNAXguEMHN10F4cAT0Gs4RmE9aHy0qBhrVzPzOOO4c4BBA@mail.gmail.com>
References: <CAE3mkNAQQYTh1wULp5ZeibMDCchz4ZFgn0caeWW-mHvHNhe8vw@mail.gmail.com>
	<CABMX_w9xB1nwtr=bcv+POkHrDwREe0ZxrM8huDOdzVLHewhKxQ@mail.gmail.com>
	<CAE3mkNAXguEMHN10F4cAT0Gs4RmE9aHy0qBhrVzPzOOO4c4BBA@mail.gmail.com>
Message-ID: <CACOYK=iyZrp1DTWZ0D58Vtc7+FAFBLG3=Ps0a3gd3wzTENm8pA@mail.gmail.com>

You need user_data.
See the example here:
https://txlab.wordpress.com/2013/06/29/freeswitch-limiting-the-number-of-concurrent-calls-on-multiple-sip-accounts/
On 3 Feb 2016 19:16, "vfclists ." <vfclists at gmail.com> wrote:

> There are quite a number of variables there, but I can't seem to find tthe
> one I want because the actual extension doesn't seem to be used in the
> configuration files.
>
> eg. When I check the logs I see output like this  sofia.c:1192
> sofia/internal/2048 at 201.182.29.148 Update Caller ID to "Desk 12" <1012>
>
> What I want to do is to combine the 2048 with the 1012 to create the
> effective_caller_id_number 20481012. The information is present somewhere
> in Freeswitch. The extensions connect to Asterisk because most of the
> applications were written for Asterisk, but Asterisk proved to be terrible
> behind NAT so a Freeswitch system was added as an intermediate gateway, so
> none of the extensions register to the Freeswitch, only the Asterisk.
>
>
> On 3 February 2016 at 17:34, ?talo Rossi <italo at freeswitch.org> wrote:
>
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/XML+Dialplan#XMLDialplan-CallerProfileFieldsvs.ChannelVariables
>>
>> On Wed, Feb 3, 2016 at 2:11 PM, vfclists . <vfclists at gmail.com> wrote:
>>
>>> What is the syntax for accessing caller profile fields in the dialplan,
>>> for instance if you want to set a channel variable to a caller profile
>>> field?
>>>
>>> Is there also a way of performing some string manipulation and
>>> extraction on the variables in the XML dialplan?
>>>
>>> This is what I am trying to achieve. The CDR from a service provider
>>> contains the CLI from a gateway on the customers premises, but it doesn't
>>> show which extension on the customer's premises the call came from. What I
>>> need is to be able to obtain the extension of the sip device and combine
>>> that with the customer gateway's CLI so that it shows in the CDR record.
>>>
>>> eg if the gateway's CLI is 2340 and the extension of the caller is 1001,
>>> corresponding to the UserId on a sipura, I want the CLI passed to the
>>> service provider to be 23401001. When I check the XML cdr in Freeswitch I
>>> see an XML value which in XPath would be accessed as
>>> /callflow/caller_profile/caller_id_number. All the information is therefore
>>> passed onto Freeswitch for use in the call, and what I need is to be able
>>> to access it and change the effective_caller_id_number before bridging the
>>> call.
>>>
>>> --
>>> 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
>>>
>>
>>
>>
>> --
>> ?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
>>
>
>
>
> --
> 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 victor.chukalovskiy at gmail.com  Wed Feb  3 23:07:29 2016
From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy)
Date: Wed, 3 Feb 2016 15:07:29 -0500
Subject: [Freeswitch-users] Is there a way to deal with "false ring-back
	than 503" carriers?
Message-ID: <56B25E01.6030608@gmail.com>

Hello,

A downstream carrier would accept INVITE, send 183 w/SDP, and almost 
instantaneously respond "503". This kills the call not letting re-route 
to the next carrier.

Without calling any specific names, this seems to become more common. 
Unfortunately, often the offending carrier is a few steps down the call 
path and I can't reach them with a cluebat.

Is there a way to make FS not fail the call and try the next route if 
using LCR as per below? Note that we are in bypass_media mode:

<action application="set" data="bypass_media=true"/>
<action application="bridge" data="lcr/lcr_profile/${destination_number}"/>

Or to keep it simple, forget LCR for a moment, can we make it go to 
gateway_2 in the following example if gw1 returns "183" followed by "503"?

<action application="set" data="bypass_media=true"/>
<action application="bridge" 
data="sofia/gateway/gw_1/${destination_number}|sofia/gateway/gw_2/${destination_number}"/>


Many thanks,
-Victor


From vfclists at gmail.com  Thu Feb  4 00:11:29 2016
From: vfclists at gmail.com (vfclists .)
Date: Wed, 3 Feb 2016 21:11:29 +0000
Subject: [Freeswitch-users] How do you access caller profile fields in
 the XML dialplan?
In-Reply-To: <CACOYK=iyZrp1DTWZ0D58Vtc7+FAFBLG3=Ps0a3gd3wzTENm8pA@mail.gmail.com>
References: <CAE3mkNAQQYTh1wULp5ZeibMDCchz4ZFgn0caeWW-mHvHNhe8vw@mail.gmail.com>
	<CABMX_w9xB1nwtr=bcv+POkHrDwREe0ZxrM8huDOdzVLHewhKxQ@mail.gmail.com>
	<CAE3mkNAXguEMHN10F4cAT0Gs4RmE9aHy0qBhrVzPzOOO4c4BBA@mail.gmail.com>
	<CACOYK=iyZrp1DTWZ0D58Vtc7+FAFBLG3=Ps0a3gd3wzTENm8pA@mail.gmail.com>
Message-ID: <CAE3mkNCU0Uydk9AMsOq9iUkLxsXB=FeqKwNc+WysShhCAFKJew@mail.gmail.com>

This is where the problem is. The place where I need that information is on
the external gateway itself, not on the internal one (behind NAT) where the
devices register. The directory settings on the internal gateway has the
effective_caller_id_name and number changed from their defaults, so the SIP
user name and user id on the sip phone don't seem to appear in any of the
channel variables during the call.

To give more detail. All SIP phone registers to Asterisk user id and
password. Asterisk registers to internal Freeswitch running on a different
port on the same box. The internal Freeswitch connects to external
Freeswitch which connects external providers gateways. I need to combine
the internal Gateway's CLI with the User ID from Sipura in the external
Freeswitch's to form the effective_caller_id_number to differentiate calls
in the external providers CDR.

The username and user id on the Sipura are present somewhere but they don't
seem to be available in any of the caller profile fields or variables in
the A Leg phase. It is in the B Leg that somehow they are obtained, and
they appear in info command, and in the Bleg xml_cdr. Can user_data
actually retrieve them in the ALeg stage?

On 3 February 2016 at 19:57, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

> You need user_data.
> See the example here:
> https://txlab.wordpress.com/2013/06/29/freeswitch-limiting-the-number-of-concurrent-calls-on-multiple-sip-accounts/
> On 3 Feb 2016 19:16, "vfclists ." <vfclists at gmail.com> wrote:
>
>> There are quite a number of variables there, but I can't seem to find
>> tthe one I want because the actual extension doesn't seem to be used in the
>> configuration files.
>>
>> eg. When I check the logs I see output like this  sofia.c:1192
>> sofia/internal/2048 at 201.182.29.148 Update Caller ID to "Desk 12" <1012>
>>
>> What I want to do is to combine the 2048 with the 1012 to create the
>> effective_caller_id_number 20481012. The information is present somewhere
>> in Freeswitch. The extensions connect to Asterisk because most of the
>> applications were written for Asterisk, but Asterisk proved to be terrible
>> behind NAT so a Freeswitch system was added as an intermediate gateway, so
>> none of the extensions register to the Freeswitch, only the Asterisk.
>>
>>
>> On 3 February 2016 at 17:34, ?talo Rossi <italo at freeswitch.org> wrote:
>>
>>>
>>> https://freeswitch.org/confluence/display/FREESWITCH/XML+Dialplan#XMLDialplan-CallerProfileFieldsvs.ChannelVariables
>>>
>>> On Wed, Feb 3, 2016 at 2:11 PM, vfclists . <vfclists at gmail.com> wrote:
>>>
>>>> What is the syntax for accessing caller profile fields in the dialplan,
>>>> for instance if you want to set a channel variable to a caller profile
>>>> field?
>>>>
>>>> Is there also a way of performing some string manipulation and
>>>> extraction on the variables in the XML dialplan?
>>>>
>>>> This is what I am trying to achieve. The CDR from a service provider
>>>> contains the CLI from a gateway on the customers premises, but it doesn't
>>>> show which extension on the customer's premises the call came from. What I
>>>> need is to be able to obtain the extension of the sip device and combine
>>>> that with the customer gateway's CLI so that it shows in the CDR record.
>>>>
>>>> eg if the gateway's CLI is 2340 and the extension of the caller is
>>>> 1001, corresponding to the UserId on a sipura, I want the CLI passed to the
>>>> service provider to be 23401001. When I check the XML cdr in Freeswitch I
>>>> see an XML value which in XPath would be accessed as
>>>> /callflow/caller_profile/caller_id_number. All the information is therefore
>>>> passed onto Freeswitch for use in the call, and what I need is to be able
>>>> to access it and change the effective_caller_id_number before bridging the
>>>> call.
>>>>
>>>> --
>>>> 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
>>>>
>>>
>>>
>>>
>>> --
>>> ?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
>>>
>>
>>
>>
>> --
>> 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
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From sos at sokhapkin.dyndns.org  Thu Feb  4 00:48:17 2016
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 03 Feb 2016 16:48:17 -0500
Subject: [Freeswitch-users] Is there a way to deal with "false ring-back
	than 503" carriers?
In-Reply-To: <56B25E01.6030608@gmail.com>
References: <56B25E01.6030608@gmail.com>
Message-ID: <1505680.nGOqxEsikt@sos>

You can't continue in bypass media mode because (wrong) media IP/port already 
sent to your downstream. Even if the second gateway completes the call OK, you 
will get one way audio.

So either proxy the media or fail the call.

On Wednesday 03 February 2016 15:07:29 Victor Chukalovskiy wrote:
> Hello,
> 
> A downstream carrier would accept INVITE, send 183 w/SDP, and almost
> instantaneously respond "503". This kills the call not letting re-route
> to the next carrier.
> 
> Without calling any specific names, this seems to become more common.
> Unfortunately, often the offending carrier is a few steps down the call
> path and I can't reach them with a cluebat.
> 
> Is there a way to make FS not fail the call and try the next route if
> using LCR as per below? Note that we are in bypass_media mode:
> 
> <action application="set" data="bypass_media=true"/>
> <action application="bridge" data="lcr/lcr_profile/${destination_number}"/>
> 
> Or to keep it simple, forget LCR for a moment, can we make it go to
> gateway_2 in the following example if gw1 returns "183" followed by "503"?
> 
> <action application="set" data="bypass_media=true"/>
> <action application="bridge"
> data="sofia/gateway/gw_1/${destination_number}|sofia/gateway/gw_2/${destinat
> ion_number}"/>
> 
> 
> Many thanks,
> -Victor
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 victor.chukalovskiy at gmail.com  Thu Feb  4 00:58:09 2016
From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy)
Date: Wed, 3 Feb 2016 16:58:09 -0500
Subject: [Freeswitch-users] Is there a way to deal with "false ring-back
 than 503" carriers?
In-Reply-To: <1505680.nGOqxEsikt@sos>
References: <56B25E01.6030608@gmail.com> <1505680.nGOqxEsikt@sos>
Message-ID: <56B277F1.5080906@gmail.com>

Thank you! Hypothesising there is a way to ontinue to #2 while in 
bypass_media, as soon as I get 183 SDP from the 2nd carrier, wouldn't it 
override bogus 183 SDP from the first carrier? It's not an ideal 
scenario to have two distinct 183's on the same call, but should be 
doable, no?

In proxy_media, is there an additional param or var I'd need in order to 
continue to #2? Would that be ignore_early_media, or something else?


On 16-02-03 04:48 PM, Sergey Okhapkin wrote:
> You can't continue in bypass media mode because (wrong) media IP/port already
> sent to your downstream. Even if the second gateway completes the call OK, you
> will get one way audio.
>
> So either proxy the media or fail the call.
>
> On Wednesday 03 February 2016 15:07:29 Victor Chukalovskiy wrote:
>> Hello,
>>
>> A downstream carrier would accept INVITE, send 183 w/SDP, and almost
>> instantaneously respond "503". This kills the call not letting re-route
>> to the next carrier.
>>
>> Without calling any specific names, this seems to become more common.
>> Unfortunately, often the offending carrier is a few steps down the call
>> path and I can't reach them with a cluebat.
>>
>> Is there a way to make FS not fail the call and try the next route if
>> using LCR as per below? Note that we are in bypass_media mode:
>>
>> <action application="set" data="bypass_media=true"/>
>> <action application="bridge" data="lcr/lcr_profile/${destination_number}"/>
>>
>> Or to keep it simple, forget LCR for a moment, can we make it go to
>> gateway_2 in the following example if gw1 returns "183" followed by "503"?
>>
>> <action application="set" data="bypass_media=true"/>
>> <action application="bridge"
>> data="sofia/gateway/gw_1/${destination_number}|sofia/gateway/gw_2/${destinat
>> ion_number}"/>
>>
>>
>> Many thanks,
>> -Victor
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 sos at sokhapkin.dyndns.org  Thu Feb  4 01:05:37 2016
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 03 Feb 2016 17:05:37 -0500
Subject: [Freeswitch-users] Is there a way to deal with "false ring-back
	than 503" carriers?
In-Reply-To: <56B277F1.5080906@gmail.com>
References: <56B25E01.6030608@gmail.com> <1505680.nGOqxEsikt@sos>
	<56B277F1.5080906@gmail.com>
Message-ID: <3151009.2tZGWi1qvm@sos>

Unfortunately 183 from the second gateway will not override the first one. It's 
how SIP offer/answer works. Once media IP provided to calling party, it can't 
be changed.

Set "continue_on_fail=true" and do not set bypass_media.

On Wednesday 03 February 2016 16:58:09 Victor Chukalovskiy wrote:
> Thank you! Hypothesising there is a way to ontinue to #2 while in
> bypass_media, as soon as I get 183 SDP from the 2nd carrier, wouldn't it
> override bogus 183 SDP from the first carrier? It's not an ideal
> scenario to have two distinct 183's on the same call, but should be
> doable, no?
> 
> In proxy_media, is there an additional param or var I'd need in order to
> continue to #2? Would that be ignore_early_media, or something else?
> 
> On 16-02-03 04:48 PM, Sergey Okhapkin wrote:
> > You can't continue in bypass media mode because (wrong) media IP/port
> > already sent to your downstream. Even if the second gateway completes the
> > call OK, you will get one way audio.
> > 
> > So either proxy the media or fail the call.
> > 
> > On Wednesday 03 February 2016 15:07:29 Victor Chukalovskiy wrote:
> >> Hello,
> >> 
> >> A downstream carrier would accept INVITE, send 183 w/SDP, and almost
> >> instantaneously respond "503". This kills the call not letting re-route
> >> to the next carrier.
> >> 
> >> Without calling any specific names, this seems to become more common.
> >> Unfortunately, often the offending carrier is a few steps down the call
> >> path and I can't reach them with a cluebat.
> >> 
> >> Is there a way to make FS not fail the call and try the next route if
> >> using LCR as per below? Note that we are in bypass_media mode:
> >> 
> >> <action application="set" data="bypass_media=true"/>
> >> <action application="bridge"
> >> data="lcr/lcr_profile/${destination_number}"/>
> >> 
> >> Or to keep it simple, forget LCR for a moment, can we make it go to
> >> gateway_2 in the following example if gw1 returns "183" followed by
> >> "503"?
> >> 
> >> <action application="set" data="bypass_media=true"/>
> >> <action application="bridge"
> >> data="sofia/gateway/gw_1/${destination_number}|sofia/gateway/gw_2/${desti
> >> nat ion_number}"/>
> >> 
> >> 
> >> Many thanks,
> >> -Victor
> >> 
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >> 
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >> 
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> > 
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 blackc2004 at gmail.com  Thu Feb  4 01:13:22 2016
From: blackc2004 at gmail.com (Cj B)
Date: Wed, 3 Feb 2016 14:13:22 -0800
Subject: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load correctly
	with DSN
Message-ID: <CAAKko=EsO-E65PK-p69jtWiuR7We0eokz0V7TU--MWEgQNp84A@mail.gmail.com>

Hi,

I've been having a very strange issue lately with loading the core codecs
when using core-db-dsn on 1.6.6 and 1.7 on Debian.

As long as I am using sqlite, everything appears to work correctly but once
I switch to using postgres the core codecs load but don't appear in the
show codecs.

uname -a
Linux c.mydomain.com 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt20-1+deb8u1
(2015-12-14) x86_64 GNU/Linux

Freeswitch start: http://pastebin.com/jD1GLFAF
It appears that they load

Show codecs after start: http://pastebin.com/W1VfsQDr
And the Core codecs are missing

Reload the core_pcm_module and show codecs: http://pastebin.com/qNpD0Due
And you can see that they are now there.

Any help here would be greatly appreciated. Hopefully I'm just missing
something small in my setups.
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From krice at freeswitch.org  Thu Feb  4 02:02:32 2016
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 3 Feb 2016 17:02:32 -0600
Subject: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load
	correctly	with DSN
In-Reply-To: <CAAKko=EsO-E65PK-p69jtWiuR7We0eokz0V7TU--MWEgQNp84A@mail.gmail.com>
References: <CAAKko=EsO-E65PK-p69jtWiuR7We0eokz0V7TU--MWEgQNp84A@mail.gmail.com>
Message-ID: <5ca701d15ed6$f7596be0$e60c43a0$@freeswitch.org>

Are you sure they aren?t actually loading? Show commands only show you whats in the database?

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Cj B
Sent: Wednesday, February 3, 2016 4:13 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load correctly with DSN

 

Hi, 

 

I've been having a very strange issue lately with loading the core codecs when using core-db-dsn on 1.6.6 and 1.7 on Debian.

 

As long as I am using sqlite, everything appears to work correctly but once I switch to using postgres the core codecs load but don't appear in the show codecs.

 

uname -a

Linux c.mydomain.com <http://c.mydomain.com>  3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt20-1+deb8u1 (2015-12-14) x86_64 GNU/Linux

 

Freeswitch start: http://pastebin.com/jD1GLFAF

It appears that they load

 

Show codecs after start: http://pastebin.com/W1VfsQDr

And the Core codecs are missing

 

Reload the core_pcm_module and show codecs: http://pastebin.com/qNpD0Due

And you can see that they are now there.

 

Any help here would be greatly appreciated. Hopefully I'm just missing something small in my setups. 

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From blackc2004 at gmail.com  Thu Feb  4 02:06:49 2016
From: blackc2004 at gmail.com (Cj B)
Date: Wed, 3 Feb 2016 15:06:49 -0800
Subject: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load
 correctly with DSN
In-Reply-To: <5ca701d15ed6$f7596be0$e60c43a0$@freeswitch.org>
References: <CAAKko=EsO-E65PK-p69jtWiuR7We0eokz0V7TU--MWEgQNp84A@mail.gmail.com>
	<5ca701d15ed6$f7596be0$e60c43a0$@freeswitch.org>
Message-ID: <CAAKko=HwNu=TUma_gJ4SuuzQveVfpY8uo0rx=PfT2oYXJX8j5A@mail.gmail.com>

On Wed, Feb 3, 2016 at 3:02 PM, Ken Rice <krice at freeswitch.org> wrote:

> Are you sure they aren?t actually loading? Show commands only show you
> whats in the database?
>
>
Hi Ken, How can I confirm that? When I try to place a call before running
the reload command  I get an error about [CS_CONSUME_MEDIA]
[INCOMPATIBLE_DESTINATION] after reloading the the calls complete, so it
seems like they are either not loading properly or they are loading before
the database?
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From krice at freeswitch.org  Thu Feb  4 02:30:52 2016
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 3 Feb 2016 17:30:52 -0600
Subject: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load
	correctly with DSN
In-Reply-To: <CAAKko=HwNu=TUma_gJ4SuuzQveVfpY8uo0rx=PfT2oYXJX8j5A@mail.gmail.com>
References: <CAAKko=EsO-E65PK-p69jtWiuR7We0eokz0V7TU--MWEgQNp84A@mail.gmail.com>	<5ca701d15ed6$f7596be0$e60c43a0$@freeswitch.org>
	<CAAKko=HwNu=TUma_gJ4SuuzQveVfpY8uo0rx=PfT2oYXJX8j5A@mail.gmail.com>
Message-ID: <5cbe01d15eda$ecb39a90$c61acfb0$@freeswitch.org>

The show commands only do a select on the database and print that out to screen? all the show commands work this way? 

 

Sounds like you have something delaying the start of the coredb?  are you doing something like trying to load configs for the core from xml_curl?

 

As far as why are you getting incompatible destination during media negotiation, post a FULL unedited call log with sip trace enabled of this scenario to the paste bin or ask on IRC someone to review your call log? the underlying error should be readily appearent there

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Cj B
Sent: Wednesday, February 3, 2016 5:07 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load correctly with DSN

 

 

On Wed, Feb 3, 2016 at 3:02 PM, Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org> > wrote:

Are you sure they aren?t actually loading? Show commands only show you whats in the database?

 

Hi Ken, How can I confirm that? When I try to place a call before running the reload command  I get an error about [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] after reloading the the calls complete, so it seems like they are either not loading properly or they are loading before the database?

 

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From max at nysolutions.com  Thu Feb  4 04:46:27 2016
From: max at nysolutions.com (Moishe Grunstein)
Date: Thu, 4 Feb 2016 01:46:27 +0000
Subject: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to
 load	correctly with DSN
In-Reply-To: <5cbe01d15eda$ecb39a90$c61acfb0$@freeswitch.org>
References: <CAAKko=EsO-E65PK-p69jtWiuR7We0eokz0V7TU--MWEgQNp84A@mail.gmail.com>
	<5ca701d15ed6$f7596be0$e60c43a0$@freeswitch.org>
	<CAAKko=HwNu=TUma_gJ4SuuzQveVfpY8uo0rx=PfT2oYXJX8j5A@mail.gmail.com>
	<5cbe01d15eda$ecb39a90$c61acfb0$@freeswitch.org>
Message-ID: <1964d5f4fef0424583e184a4b030635b@nysolutions.com>

I saw someone in the Fusion IRC channel with the same issue they said it was OK after doing a reload CORE_PCM_MODULE

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 Ken Rice
Sent: Wednesday, February 3, 2016 6:31 PM
To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load correctly with DSN

The show commands only do a select on the database and print that out to screen? all the show commands work this way?

Sounds like you have something delaying the start of the coredb?  are you doing something like trying to load configs for the core from xml_curl?

As far as why are you getting incompatible destination during media negotiation, post a FULL unedited call log with sip trace enabled of this scenario to the paste bin or ask on IRC someone to review your call log? the underlying error should be readily appearent there


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 Cj B
Sent: Wednesday, February 3, 2016 5:07 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load correctly with DSN


On Wed, Feb 3, 2016 at 3:02 PM, Ken Rice <krice at freeswitch.org<mailto:krice at freeswitch.org>> wrote:
Are you sure they aren?t actually loading? Show commands only show you whats in the database?

Hi Ken, How can I confirm that? When I try to place a call before running the reload command  I get an error about [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] after reloading the the calls complete, so it seems like they are either not loading properly or they are loading before the database?

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From blackc2004 at gmail.com  Thu Feb  4 05:48:06 2016
From: blackc2004 at gmail.com (Cj B)
Date: Wed, 3 Feb 2016 18:48:06 -0800
Subject: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load
	correctly with DSN
In-Reply-To: <5cbe01d15eda$ecb39a90$c61acfb0$@freeswitch.org>
References: <CAAKko=EsO-E65PK-p69jtWiuR7We0eokz0V7TU--MWEgQNp84A@mail.gmail.com>
	<5ca701d15ed6$f7596be0$e60c43a0$@freeswitch.org>
	<CAAKko=HwNu=TUma_gJ4SuuzQveVfpY8uo0rx=PfT2oYXJX8j5A@mail.gmail.com>
	<5cbe01d15eda$ecb39a90$c61acfb0$@freeswitch.org>
Message-ID: <68F25448-68E7-4FD7-A6BE-613DDE2D4A2A@gmail.com>

I?m using FusionPBX so I?m not sure exactly how it?s loading the configs. Here?s an unedited call log of the call: http://pastebin.com/nu3964ZA <http://pastebin.com/nu3964ZA>

You are right though that even though show codecs isn?t displaying the codecs, it appears to at least be matching them from the phone. Now that you?ve pointed that out, it?s looking more like it?s not converting the call to PCMU/PCMA for the provider? 

Thanks

> On Feb 3, 2016, at 3:30 PM, Ken Rice <krice at freeswitch.org> wrote:
> 
> The show commands only do a select on the database and print that out to screen? all the show commands work this way? 
>  
> Sounds like you have something delaying the start of the coredb?  are you doing something like trying to load configs for the core from xml_curl?
>  
> As far as why are you getting incompatible destination during media negotiation, post a FULL unedited call log with sip trace enabled of this scenario to the paste bin or ask on IRC someone to review your call log? the underlying error should be readily appearent there
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Cj B
> Sent: Wednesday, February 3, 2016 5:07 PM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load correctly with DSN
>  
>  
> On Wed, Feb 3, 2016 at 3:02 PM, Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org>> wrote:
>> Are you sure they aren?t actually loading? Show commands only show you whats in the database?
>  
> Hi Ken, How can I confirm that? When I try to place a call before running the reload command  I get an error about [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] after reloading the the calls complete, so it seems like they are either not loading properly or they are loading before the database?
>  
> _________________________________________________________________________
> 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 vfclists at gmail.com  Thu Feb  4 13:37:50 2016
From: vfclists at gmail.com (vfclists .)
Date: Thu, 4 Feb 2016 10:37:50 +0000
Subject: [Freeswitch-users] How do you check the settings a Freeswitch
	session was started with?
Message-ID: <CAE3mkNAefFPYXMTWoKSK9MMp3zRL5z=jqu3_11WGqFGiYu4upA@mail.gmail.com>

My Freeswitch is configured to start at runtime, but doesn't seem to work
until I stop it, switch to the /usr/local/freeswitch/conf directory, and
start it using '/usr/local/freeswitch/bin/freeswitch -nc'.

When I copy those commands to '/etc/rc.local', it still doesn't work ie

cd /usr/local/freeswitch/conf
/usr/local/freeswitch/bin/freeswitch -nc

I think the service script running it is not configured correctly.

Is there some fs_cli, or fsctl option that can be used to interrogate a
running instance to see what settings it was started with, or what
configuration files it is using?

-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From bote_radio at botecomm.com  Thu Feb  4 17:59:12 2016
From: bote_radio at botecomm.com (Bote Man)
Date: Thu, 4 Feb 2016 09:59:12 -0500
Subject: [Freeswitch-users] How do you check the settings a
	Freeswitch	session was started with?
In-Reply-To: <CAE3mkNAefFPYXMTWoKSK9MMp3zRL5z=jqu3_11WGqFGiYu4upA@mail.gmail.com>
References: <CAE3mkNAefFPYXMTWoKSK9MMp3zRL5z=jqu3_11WGqFGiYu4upA@mail.gmail.com>
Message-ID: <00db01d15f5c$9bef5b80$d3ce1280$@botecomm.com>

The world is moving to ?systemd? to initialize daemons and applications.

 

There are some notes about systemd on this deprecated Confluence page, but we really should figure out a dedicated documentation of systemd. Probably from the Stash source tree, I?m thinking, just to put it out there.

 

https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video

 

If you?re using sysVinit then maybe Google will turn up an older version of an init script. I think you should be looking in the o.s. logs to see what it thinks is happening when it starts FS. I think Debian stores this stuff in /var/log/access or /var/log/process.

 

---

Bote

FreeSWITCH Docs Janitor

 <http://freeswitch.org/confluence> http://freeswitch.org/confluence

 

 

 

 

From: vfclists .
Sent: Thursday, 04 February, 2016 05:38
Subject: [Freeswitch-users] How do you check the settings a Freeswitch session was started with?

 




My Freeswitch is configured to start at runtime, but doesn't seem to work until I stop it, switch to the /usr/local/freeswitch/conf directory, and start it using '/usr/local/freeswitch/bin/freeswitch -nc'.

When I copy those commands to '/etc/rc.local', it still doesn't work ie

cd /usr/local/freeswitch/conf
/usr/local/freeswitch/bin/freeswitch -nc

I think the service script running it is not configured correctly.

Is there some fs_cli, or fsctl option that can be used to interrogate a running instance to see what settings it was started with, or what configuration files it is using?

 

-- 

Frank Church

=======================
http://devblog.brahmancreations.com

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From decipher.hk at gmail.com  Thu Feb  4 03:35:12 2016
From: decipher.hk at gmail.com (=?utf-8?B?Um9kcmlnbyBSYW3DrXJleiBOb3JhbWJ1ZW5h?=)
Date: Thu, 04 Feb 2016 00:35:12 +0000
Subject: [Freeswitch-users] Answered/abandoned calls mod_callcenter
Message-ID: <701581496dd4988b32ee3289cc2166df@mail2.boxtub.com>

Hello everyone!,

I'm testing a mod_callcenter for make a FreeSWITCH version of a open source software
(https://github.com/roramirez/qpanel/tree/fs)

Now i using ESL to send command and get information from the module. I looking for a way to know the
data of answered/abandoned call from a queue and agents.

Somebody can give a tips or a light?

Regards,
--
Rodrigo Ram?rez Norambuena
http://www.rodrigoramirez.com


From tim.compnetwork at gmail.com  Thu Feb  4 19:19:18 2016
From: tim.compnetwork at gmail.com (Tim King)
Date: Thu, 4 Feb 2016 11:19:18 -0500
Subject: [Freeswitch-users] X-Auth-IP Variable?
Message-ID: <CAAKbpsW9AS2jH+3Y5tQpytnrdSGbCkk_V6YewQ1s=HYs0Yi=ww@mail.gmail.com>

I am using proxy authentication in my setup and it is working. To do this I
have created an acl
*autoload_configs/acl.conf.xml*
   <list name="opensips" default="deny">
                <node type="allow" cidr="1.2.3.4/32"/>
                <node type="allow" cidr="1.2.3.5/32"/>
                <node type="allow" cidr="1.2.3.6/32"/>
   </list>

*sip_profiles/external.xml*
<param name="auth-calls" value="true"/>
<param name="apply-proxy-acl" value="1.2.3.2/32"/>
<param name="apply-inbound-acl" value="opensips"/>


This is all working as desired. The problem is prior to adding the opensips
I was using the network_addr variable in my dialplan.

        <extension name="tollfree_to_providerA">
                <condition field="${acl(${network_addr} opensips)}"
expression="true" break="on-false"/>
                        <condition field="destination_number"
expression="^(\+?1)?(8(00|44|55|66|77|88)[2-9]\d{6})$">
                                <action application = "set"
data="dtmf_type=rfc2833"/>
                                <action application="set"
data="accountcode=customer1123"/>
                                <action application="set"
data="continue_on_fail=false"/>
                                <action application="set"
data="hangup_after_bridge=true"/>
                                <action application="set"
data="proxy_media=true"/>
                                <action application="bridge"
data="sofia/external/$2 at 8.7.6.5:5060"/>
                </condition>
        </extension>

This of course no longer works because network_addr is always the address
of my proxy server. How can I get the address from the X-authip into the
dialplan?
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From roman at dissauer.net  Thu Feb  4 19:27:18 2016
From: roman at dissauer.net (Roman Dissauer)
Date: Thu, 4 Feb 2016 17:27:18 +0100
Subject: [Freeswitch-users] BLF for Gateway
In-Reply-To: <B96DFBB9-05C1-4A14-9297-B8D2D80ABD9B@jerris.com>
References: <BD20678C-416F-413D-AC6E-077C888C3745@dissauer.net>
	<B96DFBB9-05C1-4A14-9297-B8D2D80ABD9B@jerris.com>
Message-ID: <7A4B9B3A-ABCB-493C-A9D3-CB871EE6CC73@dissauer.net>

Thanks, I?ll investigate on that and post my findings.

Best Regards,
Roman

> Am 03.02.2016 um 17:45 schrieb Michael Jerris <mike at jerris.com>:
> 
> You might be able to do something here with display updates.
> 
>> On Feb 2, 2016, at 6:58 AM, Roman Dissauer <roman at dissauer.net> wrote:
>> 
>> Hi All,
>> 
>> is there a way to get gateway usage in freeswitch on my phones blf?
>> I have multiple gateways registered and want to see which one is taken for a particular outbound call.
>> 
>> Best Regards,
>> Roman
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 andrew at cassidywebservices.co.uk  Thu Feb  4 19:44:43 2016
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Thu, 4 Feb 2016 16:44:43 +0000
Subject: [Freeswitch-users] X-Auth-IP Variable?
In-Reply-To: <CAAKbpsW9AS2jH+3Y5tQpytnrdSGbCkk_V6YewQ1s=HYs0Yi=ww@mail.gmail.com>
References: <CAAKbpsW9AS2jH+3Y5tQpytnrdSGbCkk_V6YewQ1s=HYs0Yi=ww@mail.gmail.com>
Message-ID: <CAMbW9oJSoijcU6_CMDzhsJEV_nVjmZPTvXoNVXQF03CKUYtEWw@mail.gmail.com>

${variable_sip_h_X-Auth-IP}

On 4 February 2016 at 16:19, Tim King <tim.compnetwork at gmail.com> wrote:

> I am using proxy authentication in my setup and it is working. To do this
> I have created an acl
> *autoload_configs/acl.conf.xml*
>    <list name="opensips" default="deny">
>                 <node type="allow" cidr="1.2.3.4/32"/>
>                 <node type="allow" cidr="1.2.3.5/32"/>
>                 <node type="allow" cidr="1.2.3.6/32"/>
>    </list>
>
> *sip_profiles/external.xml*
> <param name="auth-calls" value="true"/>
> <param name="apply-proxy-acl" value="1.2.3.2/32"/>
> <param name="apply-inbound-acl" value="opensips"/>
>
>
> This is all working as desired. The problem is prior to adding the
> opensips I was using the network_addr variable in my dialplan.
>
>         <extension name="tollfree_to_providerA">
>                 <condition field="${acl(${network_addr} opensips)}"
> expression="true" break="on-false"/>
>                         <condition field="destination_number"
> expression="^(\+?1)?(8(00|44|55|66|77|88)[2-9]\d{6})$">
>                                 <action application = "set"
> data="dtmf_type=rfc2833"/>
>                                 <action application="set"
> data="accountcode=customer1123"/>
>                                 <action application="set"
> data="continue_on_fail=false"/>
>                                 <action application="set"
> data="hangup_after_bridge=true"/>
>                                 <action application="set"
> data="proxy_media=true"/>
>                                 <action application="bridge"
> data="sofia/external/$2 at 8.7.6.5:5060"/>
>                 </condition>
>         </extension>
>
> This of course no longer works because network_addr is always the address
> of my proxy server. How can I get the address from the X-authip into the
> dialplan?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960  *F
<info at cassidywebservices.co.uk> *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From tim.compnetwork at gmail.com  Thu Feb  4 20:13:24 2016
From: tim.compnetwork at gmail.com (Tim King)
Date: Thu, 4 Feb 2016 12:13:24 -0500
Subject: [Freeswitch-users] X-Auth-IP Variable?
In-Reply-To: <CAMbW9oJSoijcU6_CMDzhsJEV_nVjmZPTvXoNVXQF03CKUYtEWw@mail.gmail.com>
References: <CAAKbpsW9AS2jH+3Y5tQpytnrdSGbCkk_V6YewQ1s=HYs0Yi=ww@mail.gmail.com>
	<CAMbW9oJSoijcU6_CMDzhsJEV_nVjmZPTvXoNVXQF03CKUYtEWw@mail.gmail.com>
Message-ID: <CAAKbpsVWMHMMc2i9y0XhpaTBV2gOTyiG3fDg7MrwoGz7AY8GZQ@mail.gmail.com>

Thank you for the reply. I tried this for matching to the ACL but it is
failing.
<condition field="${acl(${variable_sip_h_X-Auth-IP} opensips)}"
expression="true" break="on-false"/>

On Thu, Feb 4, 2016 at 11:44 AM, Andrew Cassidy <
andrew at cassidywebservices.co.uk> wrote:

> ${variable_sip_h_X-Auth-IP}
>
> On 4 February 2016 at 16:19, Tim King <tim.compnetwork at gmail.com> wrote:
>
>> I am using proxy authentication in my setup and it is working. To do this
>> I have created an acl
>> *autoload_configs/acl.conf.xml*
>>    <list name="opensips" default="deny">
>>                 <node type="allow" cidr="1.2.3.4/32"/>
>>                 <node type="allow" cidr="1.2.3.5/32"/>
>>                 <node type="allow" cidr="1.2.3.6/32"/>
>>    </list>
>>
>> *sip_profiles/external.xml*
>> <param name="auth-calls" value="true"/>
>> <param name="apply-proxy-acl" value="1.2.3.2/32"/>
>> <param name="apply-inbound-acl" value="opensips"/>
>>
>>
>> This is all working as desired. The problem is prior to adding the
>> opensips I was using the network_addr variable in my dialplan.
>>
>>         <extension name="tollfree_to_providerA">
>>                 <condition field="${acl(${network_addr} opensips)}"
>> expression="true" break="on-false"/>
>>                         <condition field="destination_number"
>> expression="^(\+?1)?(8(00|44|55|66|77|88)[2-9]\d{6})$">
>>                                 <action application = "set"
>> data="dtmf_type=rfc2833"/>
>>                                 <action application="set"
>> data="accountcode=customer1123"/>
>>                                 <action application="set"
>> data="continue_on_fail=false"/>
>>                                 <action application="set"
>> data="hangup_after_bridge=true"/>
>>                                 <action application="set"
>> data="proxy_media=true"/>
>>                                 <action application="bridge"
>> data="sofia/external/$2 at 8.7.6.5:5060"/>
>>                 </condition>
>>         </extension>
>>
>> This of course no longer works because network_addr is always the address
>> of my proxy server. How can I get the address from the X-authip into the
>> dialplan?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
>
>
> *T <info at cassidywebservices.co.uk> *03300 100 960  *F
> <info at cassidywebservices.co.uk> *03300 100 961
> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From andrew at cassidywebservices.co.uk  Thu Feb  4 20:23:01 2016
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Thu, 4 Feb 2016 17:23:01 +0000
Subject: [Freeswitch-users] X-Auth-IP Variable?
In-Reply-To: <CAAKbpsVWMHMMc2i9y0XhpaTBV2gOTyiG3fDg7MrwoGz7AY8GZQ@mail.gmail.com>
References: <CAAKbpsW9AS2jH+3Y5tQpytnrdSGbCkk_V6YewQ1s=HYs0Yi=ww@mail.gmail.com>
	<CAMbW9oJSoijcU6_CMDzhsJEV_nVjmZPTvXoNVXQF03CKUYtEWw@mail.gmail.com>
	<CAAKbpsVWMHMMc2i9y0XhpaTBV2gOTyiG3fDg7MrwoGz7AY8GZQ@mail.gmail.com>
Message-ID: <CAMbW9o+zGzzPLstmk-7GEHWWHXV_YC3P4A501VTfjK=OvFU0mw@mail.gmail.com>

Can you send a sip trace, particularly the one that should have the
X-Auth-IP set?

Thanks,


On 4 February 2016 at 17:13, Tim King <tim.compnetwork at gmail.com> wrote:

> Thank you for the reply. I tried this for matching to the ACL but it is
> failing.
> <condition field="${acl(${variable_sip_h_X-Auth-IP} opensips)}"
> expression="true" break="on-false"/>
>
> On Thu, Feb 4, 2016 at 11:44 AM, Andrew Cassidy <
> andrew at cassidywebservices.co.uk> wrote:
>
>> ${variable_sip_h_X-Auth-IP}
>>
>> On 4 February 2016 at 16:19, Tim King <tim.compnetwork at gmail.com> wrote:
>>
>>> I am using proxy authentication in my setup and it is working. To do
>>> this I have created an acl
>>> *autoload_configs/acl.conf.xml*
>>>    <list name="opensips" default="deny">
>>>                 <node type="allow" cidr="1.2.3.4/32"/>
>>>                 <node type="allow" cidr="1.2.3.5/32"/>
>>>                 <node type="allow" cidr="1.2.3.6/32"/>
>>>    </list>
>>>
>>> *sip_profiles/external.xml*
>>> <param name="auth-calls" value="true"/>
>>> <param name="apply-proxy-acl" value="1.2.3.2/32"/>
>>> <param name="apply-inbound-acl" value="opensips"/>
>>>
>>>
>>> This is all working as desired. The problem is prior to adding the
>>> opensips I was using the network_addr variable in my dialplan.
>>>
>>>         <extension name="tollfree_to_providerA">
>>>                 <condition field="${acl(${network_addr} opensips)}"
>>> expression="true" break="on-false"/>
>>>                         <condition field="destination_number"
>>> expression="^(\+?1)?(8(00|44|55|66|77|88)[2-9]\d{6})$">
>>>                                 <action application = "set"
>>> data="dtmf_type=rfc2833"/>
>>>                                 <action application="set"
>>> data="accountcode=customer1123"/>
>>>                                 <action application="set"
>>> data="continue_on_fail=false"/>
>>>                                 <action application="set"
>>> data="hangup_after_bridge=true"/>
>>>                                 <action application="set"
>>> data="proxy_media=true"/>
>>>                                 <action application="bridge"
>>> data="sofia/external/$2 at 8.7.6.5:5060"/>
>>>                 </condition>
>>>         </extension>
>>>
>>> This of course no longer works because network_addr is always the
>>> address of my proxy server. How can I get the address from the X-authip
>>> into the dialplan?
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>> Managing Director
>>
>>
>> *T <info at cassidywebservices.co.uk> *03300 100 960  *F
>> <info at cassidywebservices.co.uk> *03300 100 961
>> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
>> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960  *F
<info at cassidywebservices.co.uk> *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From bilaln018 at gmail.com  Thu Feb  4 20:25:00 2016
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Thu, 4 Feb 2016 22:25:00 +0500
Subject: [Freeswitch-users] [Session timer too low][SIP 422]
Message-ID: <CADSVerAVcXQCn1gEJADUXOHW04yyH_kjVxFWUbuydTaHD2tEOw@mail.gmail.com>

Hi users,
I am integrating FreeSwitch with web-sockets, using http://tryit.jssip.net/.
So my problem is the extension got registered, but after that when i dial a
number i am getting "SIP FAILURE CODE" on JsSIP, i took a network trace,
and its showing me SIP 422 responce to invite "Sesion interval too low",
now i understand that i need to increase the value of Session Expires, but
i cant increase that as it what is configured in tryjssip, so can i some
how disable this check from freeswicth?
As i guess 90 is the border value that RFC gave.

Regards
Abbasi
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From mic.niel84 at gmail.com  Thu Feb  4 20:28:05 2016
From: mic.niel84 at gmail.com (Michael Nielsen)
Date: Thu, 4 Feb 2016 18:28:05 +0100
Subject: [Freeswitch-users] loopback/XYZ doesn't contain variables from
	channel
In-Reply-To: <004401d15b96$7274e320$575ea960$@botecomm.com>
References: <CAKLKb5mjSuFapRxLkTa_HjxET0_nhDRch1jpqvgRqg26FsB7tw@mail.gmail.com>
	<CAKbxfG8crYizsAPqetGspZrhM7NVo7sbUBbyYPK6MTR94y=ypA@mail.gmail.com>
	<CAKLKb5mEUZeDuJM+8Mdw6u-1N74F-aECE++NKq3kFOnEuS5xxg@mail.gmail.com>
	<CAEwTsAjMuP=4x_mSX6u+FpeUz9ycwXvm1KzQxmvBoQhwU0zyzA@mail.gmail.com>
	<CAKLKb5nn1VX8JbcXhY0NiWpVAq7JofDNgtTNOz5jKCCQq41zLQ@mail.gmail.com>
	<CAKLKb5mTi2A4DU0x_+-A3QNzHJZR4whfY6zm59yVDq5YFJSQWw@mail.gmail.com>
	<009c01d15333$b95e5660$2c1b0320$@botecomm.com>
	<CAKLKb5kNgpj0yZxxKS2YgmLf-qvE6CPsy7tv87Xfg5PePR8+AQ@mail.gmail.com>
	<004401d15b96$7274e320$575ea960$@botecomm.com>
Message-ID: <CAKLKb5kOo3F4o80fBqRWUOeem924bnFCHWjqfNSwW_9Zi=xC9w@mail.gmail.com>

Would that be a LUA script?

On Saturday, January 30, 2016, Bote Man <bote_radio at botecomm.com> wrote:

> Perhaps your routing is getting complex enough to justify writing a script
> to do the parsing and routing? I don?t know, just suggesting that as the
> XML dialplan does have its limitations. It?s entirely possible that the XML
> dialplan can do it and I simply don?t know what it is; I am by no means a
> wizard at this.
>
>
>
> ---
>
> Bote
>
>
>
> FreeSWITCH Docs Janitor
>
> http://freeswitch.org/confluence
>
>
>
>
>
>
>
>
>
> *From:* Michael Nielsen
> *Sent:* Saturday, 30 January, 2016 03:23
> *Subject:* Re: [Freeswitch-users] loopback/XYZ doesn't contain variables
> from channel
>
>
>
> The reason for using loopback is that I'm not sure if the dialed number is
> and external or internal number.
>
> The auto dialed number is dynamic and can be everyone from 1 number to
> many numbers.
>
>
>
> Both internal and external numbers all matches real phone numbers ex.
> +44223849591.
>
>
>
> So I have another dial plan which examinate numbers and route them
> accordingly - and appends country code if necessary etc.
>
>
> On Wednesday, January 20, 2016, Bote Man <bote_radio at botecomm.com
> <javascript:_e(%7B%7D,'cvml','bote_radio at botecomm.com');>> wrote:
>
> I?m not sure that you need to use the loopback special channel to
> accomplish this. What happens if you simply set
>
>       <action application="conference_set_auto_outcall" data="user/1001"/>
>
>       <action application="conference_set_auto_outcall" data="user/1002"/>
>
>       <action application="conference_set_auto_outcall" data="3438773"/>
>
>
>
> ?and let FreeSWITCH do its normal routing? I think the loopback might be
> complicating matters. The above lines work for me, anyway.
>
>
>
> I think under the hood the auto_outcall simply stacks up originate
> commands so if the plain outside telephone number doesn?t work as-is, try
> sofia/gateway-name-here/3438773 or variations on that syntax to see if
> that works.
>
>
>
> There?s a possibility that you?re overthinking this with the loopback
> channel* J
>
>
>
> Hope this helps.
>
>
>
> * We must always keep in mind that loopback is evil. Amen.
>
>
>
> ---
>
> Bote
>
>
>
> FreeSWITCH Docs Janitor
>
> http://freeswitch.org/confluence
>
>
>
>
>
>
>
>
>
> *From:* Michael Nielsen
> *Sent:* Tuesday, 19 January, 2016 10:51
> *Subject:* Re: [Freeswitch-users] loopback/XYZ doesn't contain variables
> from channel
>
>
>
> My reason for this question is that I'm using the conference module, and
> are trying to <action application="conference_set_auto_outcall"
> data="loopback/1001,loopback/1002,loopback/3438773"/>
>
>
>
> 2 of the called numbers should be handled internally and 1 should be going
> through my sip provider.
>
> So I need all 3 of them to be handled from the top of my dial plan - with
> the correct variables set.
>
>
>
>
>
>
>
> On Fri, Jan 15, 2016 at 2:16 PM, Michael Nielsen <mic.niel84 at gmail.com>
> wrote:
>
> I've tried that as well.
>
> CLI shows:
>
> Dialplan: sofia/internal/+44234987447 at my-domain Action
> set(loopback_export=country_code)
>
> After the loopback the dialplan is as follows:
>
> Dialplan: loopback/32487477-b Action
> set(dialed_number=${country_code} 32487477) INLINE
>
> EXECUTE loopback/32487477-b set(dialed_number=32487477)
>
> So +44 is not added, even though the country_code variable is +44 before
> the loopback.
>
>
>
>
>
> On Fri, Jan 15, 2016 at 2:10 PM, Michael Jerris <mike at jerris.com> wrote:
>
>         <action application="set" data="loopback_export=country_code"/>
>
>
>
> On Friday, January 15, 2016, Michael Nielsen <mic.niel84 at gmail.com> wrote:
>
> I've tried to add:
>
>         <action application="set"
> data="loopback_export=[country_code=${country_code}]"/>
>
>
>
> And my log shows the following:
>
> set(loopback_export=[country_code=+44])
>
> But it still isn't available on the next runthrough of dial plan.
>
>
>
> On Wed, Jan 13, 2016 at 6:51 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
> set
>
> loopback_export to the names of all of the variables you want to copy
> across sep buy a , (comma)
>
>
>
>
>
> On Wed, Jan 13, 2016 at 8:23 AM, Michael Nielsen <mic.niel84 at gmail.com>
> wrote:
>
> I've got directory with subscribers containing a variable called
> "country_code".
>
> This variable can be used in regular dial plans and works perfectly.
>
>
>
> However, I've got a conference call dial plan where I
> use conference_set_auto_outcall.
>
> I'mm calling loopback/XYZ to get my other dial plans into play.
>
>
>
> Everything works, except my subscriber variable "country_code" doesn't get
> recognised after the loopback/.
>
>
>
> How can I fix this kind of issue?
>
>
>
>
>
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From bote_radio at botecomm.com  Thu Feb  4 20:45:17 2016
From: bote_radio at botecomm.com (Bote Man)
Date: Thu, 4 Feb 2016 12:45:17 -0500
Subject: [Freeswitch-users] loopback/XYZ doesn't contain variables
	from	channel
In-Reply-To: <CAKLKb5kOo3F4o80fBqRWUOeem924bnFCHWjqfNSwW_9Zi=xC9w@mail.gmail.com>
References: <CAKLKb5mjSuFapRxLkTa_HjxET0_nhDRch1jpqvgRqg26FsB7tw@mail.gmail.com>	<CAKbxfG8crYizsAPqetGspZrhM7NVo7sbUBbyYPK6MTR94y=ypA@mail.gmail.com>	<CAKLKb5mEUZeDuJM+8Mdw6u-1N74F-aECE++NKq3kFOnEuS5xxg@mail.gmail.com>	<CAEwTsAjMuP=4x_mSX6u+FpeUz9ycwXvm1KzQxmvBoQhwU0zyzA@mail.gmail.com>	<CAKLKb5nn1VX8JbcXhY0NiWpVAq7JofDNgtTNOz5jKCCQq41zLQ@mail.gmail.com>	<CAKLKb5mTi2A4DU0x_+-A3QNzHJZR4whfY6zm59yVDq5YFJSQWw@mail.gmail.com>	<009c01d15333$b95e5660$2c1b0320$@botecomm.com>	<CAKLKb5kNgpj0yZxxKS2YgmLf-qvE6CPsy7tv87Xfg5PePR8+AQ@mail.gmail.com>	<004401d15b96$7274e320$575ea960$@botecomm.com>
	<CAKLKb5kOo3F4o80fBqRWUOeem924bnFCHWjqfNSwW_9Zi=xC9w@mail.gmail.com>
Message-ID: <003701d15f73$cfeae2d0$6fc0a870$@botecomm.com>

Lua, perl, whatever is your strongest suit. You should not do telephony in the script, just figure out the best route to the destination and present that to FreeSWITCH to do the heavy lifting of telephony signaling and media setup.

 

This is one solution that springs to mind, but perhaps the XML dialplan can do it, I don?t know.

 

Bote

 

 

From: Michael Nielsen
Sent: Thursday, 04 February, 2016 12:28
Subject: Re: [Freeswitch-users] loopback/XYZ doesn't contain variables from channel

 

Would that be a LUA script?

On Saturday, January 30, 2016, Bote Man <bote_radio at botecomm.com> wrote:

Perhaps your routing is getting complex enough to justify writing a script to do the parsing and routing? I don?t know, just suggesting that as the XML dialplan does have its limitations. It?s entirely possible that the XML dialplan can do it and I simply don?t know what it is; I am by no means a wizard at this.

 

---

Bote

 

FreeSWITCH Docs Janitor

http://freeswitch.org/confluence

 

 

 

 

From: Michael Nielsen
Sent: Saturday, 30 January, 2016 03:23
Subject: Re: [Freeswitch-users] loopback/XYZ doesn't contain variables from channel

 

The reason for using loopback is that I'm not sure if the dialed number is and external or internal number. 

The auto dialed number is dynamic and can be everyone from 1 number to many numbers. 

 

Both internal and external numbers all matches real phone numbers ex. +44223849591.

 

So I have another dial plan which examinate numbers and route them accordingly - and appends country code if necessary etc. 


On Wednesday, January 20, 2016, Bote Man <bote_radio at botecomm.com <javascript:_e(%7B%7D,'cvml','bote_radio at botecomm.com');> > wrote:

I?m not sure that you need to use the loopback special channel to accomplish this. What happens if you simply set 

      <action application="conference_set_auto_outcall" data="user/1001"/>

      <action application="conference_set_auto_outcall" data="user/1002"/>

      <action application="conference_set_auto_outcall" data="3438773"/>

 

?and let FreeSWITCH do its normal routing? I think the loopback might be complicating matters. The above lines work for me, anyway.

 

I think under the hood the auto_outcall simply stacks up originate commands so if the plain outside telephone number doesn?t work as-is, try sofia/gateway-name-here/3438773 or variations on that syntax to see if that works.

 

There?s a possibility that you?re overthinking this with the loopback channel* J

 

Hope this helps.

 

* We must always keep in mind that loopback is evil. Amen.

 

---

Bote

 

FreeSWITCH Docs Janitor

http://freeswitch.org/confluence

 

 

 

 

From: Michael Nielsen
Sent: Tuesday, 19 January, 2016 10:51
Subject: Re: [Freeswitch-users] loopback/XYZ doesn't contain variables from channel

 

My reason for this question is that I'm using the conference module, and are trying to <action application="conference_set_auto_outcall" data="loopback/1001,loopback/1002,loopback/3438773"/>

 

2 of the called numbers should be handled internally and 1 should be going through my sip provider.

So I need all 3 of them to be handled from the top of my dial plan - with the correct variables set.

 

 

 

On Fri, Jan 15, 2016 at 2:16 PM, Michael Nielsen <mic.niel84 at gmail.com> wrote:

I've tried that as well.

CLI shows:

Dialplan: sofia/internal/+44234987447 at my-domain Action set(loopback_export=country_code) 

After the loopback the dialplan is as follows:

Dialplan: loopback/32487477-b Action set(dialed_number=${country_code} 32487477) INLINE

EXECUTE loopback/32487477-b set(dialed_number=32487477)

So +44 is not added, even though the country_code variable is +44 before the loopback.

 

 

On Fri, Jan 15, 2016 at 2:10 PM, Michael Jerris <mike at jerris.com> wrote:

        <action application="set" data="loopback_export=country_code"/>



On Friday, January 15, 2016, Michael Nielsen <mic.niel84 at gmail.com> wrote:

I've tried to add:

        <action application="set" data="loopback_export=[country_code=${country_code}]"/>

 

And my log shows the following:

set(loopback_export=[country_code=+44])

But it still isn't available on the next runthrough of dial plan.

 

On Wed, Jan 13, 2016 at 6:51 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

set  

loopback_export to the names of all of the variables you want to copy across sep buy a , (comma)

 

 

On Wed, Jan 13, 2016 at 8:23 AM, Michael Nielsen <mic.niel84 at gmail.com> wrote:

I've got directory with subscribers containing a variable called "country_code".

This variable can be used in regular dial plans and works perfectly.

 

However, I've got a conference call dial plan where I use conference_set_auto_outcall.

I'mm calling loopback/XYZ to get my other dial plans into play.

 

Everything works, except my subscriber variable "country_code" doesn't get recognised after the loopback/.

 

How can I fix this kind of issue?

 

 

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From bote_radio at botecomm.com  Thu Feb  4 20:45:17 2016
From: bote_radio at botecomm.com (Bote Man)
Date: Thu, 4 Feb 2016 12:45:17 -0500
Subject: [Freeswitch-users] X-Auth-IP Variable?
In-Reply-To: <CAAKbpsVWMHMMc2i9y0XhpaTBV2gOTyiG3fDg7MrwoGz7AY8GZQ@mail.gmail.com>
References: <CAAKbpsW9AS2jH+3Y5tQpytnrdSGbCkk_V6YewQ1s=HYs0Yi=ww@mail.gmail.com>	<CAMbW9oJSoijcU6_CMDzhsJEV_nVjmZPTvXoNVXQF03CKUYtEWw@mail.gmail.com>
	<CAAKbpsVWMHMMc2i9y0XhpaTBV2gOTyiG3fDg7MrwoGz7AY8GZQ@mail.gmail.com>
Message-ID: <003c01d15f73$d07a5190$716ef4b0$@botecomm.com>

I also recommend using the FreeSWITCH ?log? application to display what FS thinks those variables contain. 

 

 

---

Bote

FreeSWITCH Docs Janitor

 <http://freeswitch.org/confluence> http://freeswitch.org/confluence

 

 

 

 

From: Tim King
Sent: Thursday, 04 February, 2016 12:13
Subject: Re: [Freeswitch-users] X-Auth-IP Variable?

 

Thank you for the reply. I tried this for matching to the ACL but it is failing.

<condition field="${acl(${variable_sip_h_X-Auth-IP} opensips)}" expression="true" break="on-false"/>

 

On Thu, Feb 4, 2016 at 11:44 AM, Andrew Cassidy <andrew at cassidywebservices.co.uk> wrote:

${variable_sip_h_X-Auth-IP}

 

On 4 February 2016 at 16:19, Tim King <tim.compnetwork at gmail.com> wrote:

I am using proxy authentication in my setup and it is working. To do this I have created an acl

autoload_configs/acl.conf.xml

   <list name="opensips" default="deny">

                <node type="allow" cidr="1.2.3.4/32"/>

                <node type="allow" cidr="1.2.3.5/32"/>

                <node type="allow" cidr="1.2.3.6/32"/>

   </list>

 

sip_profiles/external.xml

<param name="auth-calls" value="true"/>

<param name="apply-proxy-acl" value="1.2.3.2/32"/>

<param name="apply-inbound-acl" value="opensips"/>

 

 

This is all working as desired. The problem is prior to adding the opensips I was using the network_addr variable in my dialplan.

 

        <extension name="tollfree_to_providerA">

                <condition field="${acl(${network_addr} opensips)}" expression="true" break="on-false"/>

                        <condition field="destination_number" expression="^(\+?1)?(8(00|44|55|66|77|88)[2-9]\d{6})$">

                                <action application = "set" data="dtmf_type=rfc2833"/>

                                <action application="set" data="accountcode=customer1123"/>

                                <action application="set" data="continue_on_fail=false"/>

                                <action application="set" data="hangup_after_bridge=true"/>

                                <action application="set" data="proxy_media=true"/>

                                <action application="bridge" data="sofia/external/$2 at 8.7.6.5:5060"/>

                </condition>

        </extension>

 

This of course no longer works because network_addr is always the address of my proxy server. How can I get the address from the X-authip into the dialplan?

 

 

 

-- 

Andrew Cassidy BSc (Hons) MBCS SSCA

Managing Director

  <http://www.cassidywebservices.co.uk/static/emailsig.png> 

 

 <mailto:info at cassidywebservices.co.uk> T 03300 100 960   <mailto:info at cassidywebservices.co.uk> F 03300 100 961

 <mailto:info at cassidywebservices.co.uk> E andrew at cassidywebservices.co.uk

 <mailto:info at cassidywebservices.co.uk> W www.cassidywebservices.co.uk


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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at 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 tim.compnetwork at gmail.com  Thu Feb  4 20:52:19 2016
From: tim.compnetwork at gmail.com (Tim King)
Date: Thu, 4 Feb 2016 12:52:19 -0500
Subject: [Freeswitch-users] X-Auth-IP Variable?
In-Reply-To: <003c01d15f73$d07a5190$716ef4b0$@botecomm.com>
References: <CAAKbpsW9AS2jH+3Y5tQpytnrdSGbCkk_V6YewQ1s=HYs0Yi=ww@mail.gmail.com>
	<CAMbW9oJSoijcU6_CMDzhsJEV_nVjmZPTvXoNVXQF03CKUYtEWw@mail.gmail.com>
	<CAAKbpsVWMHMMc2i9y0XhpaTBV2gOTyiG3fDg7MrwoGz7AY8GZQ@mail.gmail.com>
	<003c01d15f73$d07a5190$716ef4b0$@botecomm.com>
Message-ID: <CAAKbpsWKvTMcx_HxW7gDaGMjTfk6Ve-+-NbavXoeKY0Cb5NOdg@mail.gmail.com>

I keep trying to log them but the log application is showing me the log
statement and not the variable itself. I know I have used this successfully
before, but even if I try logging the destination_number it does not log.

I have tried ${destination_number} and [${destination_number}] and my log
output looks like this.

Action log(Log TEST ========== ${destination_number} or
[${destination_number}]

On Thu, Feb 4, 2016 at 12:45 PM, Bote Man <bote_radio at botecomm.com> wrote:

> I also recommend using the FreeSWITCH ?log? application to display what FS
> thinks those variables contain.
>
>
>
>
>
> ---
>
> Bote
>
> FreeSWITCH Docs Janitor
>
> http://freeswitch.org/confluence
>
>
>
>
>
>
>
>
>
> *From:* Tim King
> *Sent:* Thursday, 04 February, 2016 12:13
> *Subject:* Re: [Freeswitch-users] X-Auth-IP Variable?
>
>
>
> Thank you for the reply. I tried this for matching to the ACL but it is
> failing.
>
> <condition field="${acl(${variable_sip_h_X-Auth-IP} opensips)}"
> expression="true" break="on-false"/>
>
>
>
> On Thu, Feb 4, 2016 at 11:44 AM, Andrew Cassidy <
> andrew at cassidywebservices.co.uk> wrote:
>
> ${variable_sip_h_X-Auth-IP}
>
>
>
> On 4 February 2016 at 16:19, Tim King <tim.compnetwork at gmail.com> wrote:
>
> I am using proxy authentication in my setup and it is working. To do this
> I have created an acl
>
> *autoload_configs/acl.conf.xml*
>
>    <list name="opensips" default="deny">
>
>                 <node type="allow" cidr="1.2.3.4/32"/>
>
>                 <node type="allow" cidr="1.2.3.5/32"/>
>
>                 <node type="allow" cidr="1.2.3.6/32"/>
>
>    </list>
>
>
>
> *sip_profiles/external.xml*
>
> <param name="auth-calls" value="true"/>
>
> <param name="apply-proxy-acl" value="1.2.3.2/32"/>
>
> <param name="apply-inbound-acl" value="opensips"/>
>
>
>
>
>
> This is all working as desired. The problem is prior to adding the
> opensips I was using the network_addr variable in my dialplan.
>
>
>
>         <extension name="tollfree_to_providerA">
>
>                 <condition field="${acl(${network_addr} opensips)}"
> expression="true" break="on-false"/>
>
>                         <condition field="destination_number"
> expression="^(\+?1)?(8(00|44|55|66|77|88)[2-9]\d{6})$">
>
>                                 <action application = "set"
> data="dtmf_type=rfc2833"/>
>
>                                 <action application="set"
> data="accountcode=customer1123"/>
>
>                                 <action application="set"
> data="continue_on_fail=false"/>
>
>                                 <action application="set"
> data="hangup_after_bridge=true"/>
>
>                                 <action application="set"
> data="proxy_media=true"/>
>
>                                 <action application="bridge"
> data="sofia/external/$2 at 8.7.6.5:5060"/>
>
>                 </condition>
>
>         </extension>
>
>
>
> This of course no longer works because network_addr is always the address
> of my proxy server. How can I get the address from the X-authip into the
> dialplan?
>
>
>
>
>
>
>
> --
>
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>
> Managing Director
>
>
>
> *T <info at cassidywebservices.co.uk> *03300 100 960  *F
> <info at cassidywebservices.co.uk> *03300 100 961
>
> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
>
> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 blackc2004 at gmail.com  Thu Feb  4 21:02:33 2016
From: blackc2004 at gmail.com (Cj B)
Date: Thu, 4 Feb 2016 10:02:33 -0800
Subject: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load
	correctly with DSN
In-Reply-To: <68F25448-68E7-4FD7-A6BE-613DDE2D4A2A@gmail.com>
References: <CAAKko=EsO-E65PK-p69jtWiuR7We0eokz0V7TU--MWEgQNp84A@mail.gmail.com>
	<5ca701d15ed6$f7596be0$e60c43a0$@freeswitch.org>
	<CAAKko=HwNu=TUma_gJ4SuuzQveVfpY8uo0rx=PfT2oYXJX8j5A@mail.gmail.com>
	<5cbe01d15eda$ecb39a90$c61acfb0$@freeswitch.org>
	<68F25448-68E7-4FD7-A6BE-613DDE2D4A2A@gmail.com>
Message-ID: <CC5D7440-B4C4-453A-94BF-E2CC44E9276A@gmail.com>

Good Morning,

I worked with the fusion guys and found that if I set absolut_codec_string=pcma on the outbound routes the calls are completing ok. I also have inbound-codec-negotiation=generous on my sip profiles, maybe I need to change that to scrooge or something else? 

Any other suggestions?

Thanks

> On Feb 3, 2016, at 6:48 PM, Cj B <blackc2004 at gmail.com> wrote:
> 
> I?m using FusionPBX so I?m not sure exactly how it?s loading the configs. Here?s an unedited call log of the call: http://pastebin.com/nu3964ZA <http://pastebin.com/nu3964ZA>
> 
> You are right though that even though show codecs isn?t displaying the codecs, it appears to at least be matching them from the phone. Now that you?ve pointed that out, it?s looking more like it?s not converting the call to PCMU/PCMA for the provider? 
> 
> Thanks
> 
>> On Feb 3, 2016, at 3:30 PM, Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org>> wrote:
>> 
>> The show commands only do a select on the database and print that out to screen? all the show commands work this way? 
>>  
>> Sounds like you have something delaying the start of the coredb?  are you doing something like trying to load configs for the core from xml_curl?
>>  
>> As far as why are you getting incompatible destination during media negotiation, post a FULL unedited call log with sip trace enabled of this scenario to the paste bin or ask on IRC someone to review your call log? the underlying error should be readily appearent there
>>  
>>  
>> 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 Cj B
>> Sent: Wednesday, February 3, 2016 5:07 PM
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
>> Subject: Re: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load correctly with DSN
>>  
>>  
>> On Wed, Feb 3, 2016 at 3:02 PM, Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org>> wrote:
>>> Are you sure they aren?t actually loading? Show commands only show you whats in the database?
>>  
>> Hi Ken, How can I confirm that? When I try to place a call before running the reload command  I get an error about [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] after reloading the the calls complete, so it seems like they are either not loading properly or they are loading before the database?
>>  
>> _________________________________________________________________________
>> 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 brian at freeswitch.org  Thu Feb  4 21:05:39 2016
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Feb 2016 12:05:39 -0600
Subject: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load
 correctly with DSN
In-Reply-To: <CC5D7440-B4C4-453A-94BF-E2CC44E9276A@gmail.com>
References: <CAAKko=EsO-E65PK-p69jtWiuR7We0eokz0V7TU--MWEgQNp84A@mail.gmail.com>
	<5ca701d15ed6$f7596be0$e60c43a0$@freeswitch.org>
	<CAAKko=HwNu=TUma_gJ4SuuzQveVfpY8uo0rx=PfT2oYXJX8j5A@mail.gmail.com>
	<5cbe01d15eda$ecb39a90$c61acfb0$@freeswitch.org>
	<68F25448-68E7-4FD7-A6BE-613DDE2D4A2A@gmail.com>
	<CC5D7440-B4C4-453A-94BF-E2CC44E9276A@gmail.com>
Message-ID: <CAEJMVkBi2-XzY4a-OvBgeKz_hP6U683SDu5rjH+LgWhUkHw6oA@mail.gmail.com>

The codec is loading and your do config is broken during start up so it may
not appear in show modules in this case, we may need to have a JIRA filed
in this so we can investigate and get the facts

On Thursday, February 4, 2016, Cj B <blackc2004 at gmail.com> wrote:

> Good Morning,
>
> I worked with the fusion guys and found that if I set
> absolut_codec_string=pcma on the outbound routes the calls are completing
> ok. I also have inbound-codec-negotiation=generous on my sip profiles,
> maybe I need to change that to scrooge or something else?
>
> Any other suggestions?
>
> Thanks
>
> On Feb 3, 2016, at 6:48 PM, Cj B <blackc2004 at gmail.com
> <javascript:_e(%7B%7D,'cvml','blackc2004 at gmail.com');>> wrote:
>
> I?m using FusionPBX so I?m not sure exactly how it?s loading the configs.
> Here?s an unedited call log of the call: http://pastebin.com/nu3964ZA
>
> You are right though that even though show codecs isn?t displaying the
> codecs, it appears to at least be matching them from the phone. Now that
> you?ve pointed that out, it?s looking more like it?s not converting the
> call to PCMU/PCMA for the provider?
>
> Thanks
>
> On Feb 3, 2016, at 3:30 PM, Ken Rice <krice at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','krice at freeswitch.org');>> wrote:
>
> The show commands only do a select on the database and print that out to
> screen? all the show commands work this way?
>
> Sounds like you have something delaying the start of the coredb?  are you
> doing something like trying to load configs for the core from xml_curl?
>
> As far as why are you getting incompatible destination during media
> negotiation, post a FULL unedited call log with sip trace enabled of this
> scenario to the paste bin or ask on IRC someone to review your call log?
> the underlying error should be readily appearent there
>
>
> *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 *Cj B
> *Sent:* Wednesday, February 3, 2016 5:07 PM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users at lists.freeswitch.org');>>
> *Subject:* Re: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load
> correctly with DSN
>
>
> On Wed, Feb 3, 2016 at 3:02 PM, Ken Rice <krice at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','krice at freeswitch.org');>> wrote:
>
> Are you sure they aren?t actually loading? Show commands only show you
> whats in the database?
>
>
> Hi Ken, How can I confirm that? When I try to place a call before running
> the reload command  I get an error about [CS_CONSUME_MEDIA]
> [INCOMPATIBLE_DESTINATION] after reloading the the calls complete, so it
> seems like they are either not loading properly or they are loading before
> the database?
>
> _________________________________________________________________________
> 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


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

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 anthony.minessale at gmail.com  Thu Feb  4 23:08:10 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Feb 2016 14:08:10 -0600
Subject: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load
 correctly with DSN
In-Reply-To: <CAEJMVkBi2-XzY4a-OvBgeKz_hP6U683SDu5rjH+LgWhUkHw6oA@mail.gmail.com>
References: <CAAKko=EsO-E65PK-p69jtWiuR7We0eokz0V7TU--MWEgQNp84A@mail.gmail.com>
	<5ca701d15ed6$f7596be0$e60c43a0$@freeswitch.org>
	<CAAKko=HwNu=TUma_gJ4SuuzQveVfpY8uo0rx=PfT2oYXJX8j5A@mail.gmail.com>
	<5cbe01d15eda$ecb39a90$c61acfb0$@freeswitch.org>
	<68F25448-68E7-4FD7-A6BE-613DDE2D4A2A@gmail.com>
	<CC5D7440-B4C4-453A-94BF-E2CC44E9276A@gmail.com>
	<CAEJMVkBi2-XzY4a-OvBgeKz_hP6U683SDu5rjH+LgWhUkHw6oA@mail.gmail.com>
Message-ID: <CAKbxfG8Goe6Z+gu1wGKD8zQ=8-vUwzE6hQWOmO3=Fq+CX-agfQ@mail.gmail.com>

Look for errors in the SQL server logs right after a fresh restart.


On Thu, Feb 4, 2016 at 12:05 PM, Brian West <brian at freeswitch.org> wrote:

> The codec is loading and your do config is broken during start up so it
> may not appear in show modules in this case, we may need to have a JIRA
> filed in this so we can investigate and get the facts
>
>
> On Thursday, February 4, 2016, Cj B <blackc2004 at gmail.com> wrote:
>
>> Good Morning,
>>
>> I worked with the fusion guys and found that if I set
>> absolut_codec_string=pcma on the outbound routes the calls are completing
>> ok. I also have inbound-codec-negotiation=generous on my sip profiles,
>> maybe I need to change that to scrooge or something else?
>>
>> Any other suggestions?
>>
>> Thanks
>>
>> On Feb 3, 2016, at 6:48 PM, Cj B <blackc2004 at gmail.com> wrote:
>>
>> I?m using FusionPBX so I?m not sure exactly how it?s loading the configs.
>> Here?s an unedited call log of the call: http://pastebin.com/nu3964ZA
>>
>> You are right though that even though show codecs isn?t displaying the
>> codecs, it appears to at least be matching them from the phone. Now that
>> you?ve pointed that out, it?s looking more like it?s not converting the
>> call to PCMU/PCMA for the provider?
>>
>> Thanks
>>
>> On Feb 3, 2016, at 3:30 PM, Ken Rice <krice at freeswitch.org> wrote:
>>
>> The show commands only do a select on the database and print that out to
>> screen? all the show commands work this way?
>>
>> Sounds like you have something delaying the start of the coredb?  are you
>> doing something like trying to load configs for the core from xml_curl?
>>
>> As far as why are you getting incompatible destination during media
>> negotiation, post a FULL unedited call log with sip trace enabled of this
>> scenario to the paste bin or ask on IRC someone to review your call log?
>> the underlying error should be readily appearent there
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [
>> mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Cj B
>> *Sent:* Wednesday, February 3, 2016 5:07 PM
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* Re: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load
>> correctly with DSN
>>
>>
>> On Wed, Feb 3, 2016 at 3:02 PM, Ken Rice <krice at freeswitch.org> wrote:
>>
>> Are you sure they aren?t actually loading? Show commands only show you
>> whats in the database?
>>
>>
>> Hi Ken, How can I confirm that? When I try to place a call before running
>> the reload command  I get an error about [CS_CONSUME_MEDIA]
>> [INCOMPATIBLE_DESTINATION] after reloading the the calls complete, so it
>> seems like they are either not loading properly or they are loading before
>> the database?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> 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
>



-- 
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 lists at telefaks.de  Fri Feb  5 00:20:26 2016
From: lists at telefaks.de (Peter Steinbach)
Date: Thu, 04 Feb 2016 22:20:26 +0100
Subject: [Freeswitch-users] fs_cli under Windows and cygwin, no prompt
Message-ID: <56B3C09A.60907@telefaks.de>

I have inherited a Freeswitch installation on a Windows server, which I
remotely manage with cygwin and ssh.

This worked more or less nicely so far. But today I cannot manage
Freeswitch via ssh with fs_cli anymore. I restarted the hardware, but no
change.
I can run fs_cli.exe, and I see the Freeswitch messages, but I do
__not__ get a prompt. When I try to enter characters, nothing happens.

On the machine itself, when I open a console and start cygwin,
fs_cli.exe works as expected with prompt. But not via ssh. But
fs_cli.exe -x "command" works via ssh, so login to Freeswitch should be
fine in general.

What I understand is, that fs_cli connects to Freeswitch via port 8021.
So when seeing FS messages, I expect that the connection is already
fine, right?

I also tried to connect via telnet to port 8021. This works in the local
cygwin shell (get an auth request), but not via remote ssh shell (telnet
terminates without message).

I am puzzled. Anybody has a clue, why I do not have a prompt and where
to search?

-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de



From peterg at sytelco.com  Thu Feb  4 22:42:36 2016
From: peterg at sytelco.com (Piotr Gregor)
Date: Thu, 4 Feb 2016 19:42:36 +0000
Subject: [Freeswitch-users] Best practices for determining if originated
 call is answered by human or voicemail system?
In-Reply-To: <CAL7Tj2_JVak-y=x-CpGyfMnqMZTkjjDTzaT93Pm3X1h5VN0RxA@mail.gmail.com>
References: <CAAemr3yoTC5obmo_SoGcyO+AehMOedRyQHSK41_d02dZrq_teA@mail.gmail.com>
	<CAD-qRkjhTHdWBxVgm5baWSV+fLFuPyvzYW4JLf+WB_REUVRN5g@mail.gmail.com>
	<CAL7Tj2_JVak-y=x-CpGyfMnqMZTkjjDTzaT93Pm3X1h5VN0RxA@mail.gmail.com>
Message-ID: <CAL7Tj2_oKsj8dAZXTZXURcEgRDqpHsbVfdQzBCosfKWLKK529Q@mail.gmail.com>

Hi,
"I looked at the AVMD module documentation at
https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd , and it
seems to indicate that this simply detects a beep, i.e. it does not use
talking / silence heuristics into account to determine if the call is
answered by a human or machine.
Am I correct about that? "

Yes. The AVMD module works by calculating the estimate of the frequency and
amplitude of signal using DESA-2 algorithm. It fires amd::beep event when
the variance for the frequency estimate is below the threshold.
The event is of form:

Event-Subclass: avmd%3A%3Abeep
Event-Name: CUSTOM
Core-UUID: 36a5b214-8d83-487e-8f8b-50af9a090486
FreeSWITCH-Hostname: home
FreeSWITCH-Switchname: home
FreeSWITCH-IPv4: 128.11.35.8
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2016-02-04%2016%3A56%3A55
Event-Date-GMT: Thu,%2004%20Feb%202016%2016%3A56%3A55%20GMT
Event-Date-Timestamp: 1454605015915799
Event-Calling-File: mod_avmd.c
Event-Calling-Function: avmd_process
Event-Calling-Line-Number: 556
Event-Sequence: 950
Beep-Status: stop
Unique-ID: 0a6f6df3-cc48-4b8a-93f3-b35ee1b8ef0a
call-command: avmd

The AVMD doesn't take into consideration segments of speech and segments of
silence. It simply calculates it's estimate of frequency and amplitude.


"How is this intended to be used, do I assume that the call is answered by
a human until this module sends a avmd%3A%3Abeep event to me?  How do
others use this module or other techniques to determine human / machine
answer to outbound calls with Freeswitch? "

It's use is to detect beep only.
You can test how it works by creating extension that will play a tone of
given frequency, e.g:
    <extension name="10014">
      <condition field="destination_number" expression="^10014$">
        <action application="playback"
data="tone_stream://L=3;%(500,6850,850)" />
        <action application="hangup"/>
      </condition>
    </extension>
This plays tone of 500Hz frequency and 6850 ms long, then silence by 850 ms.

Start avmd on the call with fs_cli:
avmd 8c6a6bb1-ec64-4b2c-9e48-93a6a2598b64 start

and inspect events by regestering to avmd events with:
events plain CUSTOM avmd::beep

Analysing the audio for a presence of the tone is not enough for answering
machine detection. You should also analyse at least the length of
speech/silence segments. You can do this by subscription to TALK/NOTALK
events.
Hope this helps.

cheers,
Piotr

On 4 February 2016 at 17:50, Piotr Gregor <peterg at sytelco.com> wrote:

> Hi,
> "I looked at the AVMD module documentation at
> https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd , and it
> seems to indicate that this simply detects a beep, i.e. it does not use
> talking / silence heuristics into account to determine if the call is
> answered by a human or machine.
> Am I correct about that? "
>
> Yes. The AVMD module works by calculating the estimate of the frequency
> using DESA-2 algorithm. It fires amd::beep event when the variance for that
> estimate is below threshold.
> The event is of form:
>
> Event-Subclass: avmd%3A%3Abeep
> Event-Name: CUSTOM
> Core-UUID: 36a5b214-8d83-487e-8f8b-50af9a090486
> FreeSWITCH-Hostname: home
> FreeSWITCH-Switchname: home
> FreeSWITCH-IPv4: 128.11.35.8
> FreeSWITCH-IPv6: %3A%3A1
> Event-Date-Local: 2016-02-04%2016%3A56%3A55
> Event-Date-GMT: Thu,%2004%20Feb%202016%2016%3A56%3A55%20GMT
> Event-Date-Timestamp: 1454605015915799
> Event-Calling-File: mod_avmd.c
> Event-Calling-Function: avmd_process
> Event-Calling-Line-Number: 556
> Event-Sequence: 950
> Beep-Status: stop
> Unique-ID: 0a6f6df3-cc48-4b8a-93f3-b35ee1b8ef0a
> call-command: avmd
>
> The AVMD doesn't take into consideration TALK/NOTALK events.
>
>
> "How is this intended to be used, do I assume that the call is answered
> by a human until this module sends a avmd%3A%3Abeep event to me?  How do
> others use this module or other techniques to determine human / machine
> answer to outbound calls with Freeswitch? "
>
> You can test how it works by creating extension that will play a tone of
> given frequency, e.g:
>     <extension name="10014">
>       <condition field="destination_number" expression="^10014$">
>         <action application="playback"
> data="tone_stream://L=3;%(500,6850,850)" />
>         <action application="hangup"/>
>       </condition>
>     </extension>
> This plays tone of 500Hz frequency and 6850 ms long, then silence by 850
> ms.
>
> Start avmd on the call with fs_cli:
> avmd 8c6a6bb1-ec64-4b2c-9e48-93a6a2598b64 start
>
> and inspect events by regestering to avmd events with:
> events plain CUSTOM avmd::beep
>
> Analysing the audio for a presence of the tone is not enough for answering
> machine detection. You should also analyse at least the length of
> speech/silence segments.
> Hope this helps.
>
> cheers,
> Piotr
>
>
>
>
>
>
>
> On 3 February 2016 at 19:50, Christopher Rienzo <cmrienzo at gmail.com>
> wrote:
>
>> I wouldn't use beep detection in a dialer application, but it could be
>> useful in something like follow me to reduce the occurrence of voicemails
>> being left on subscriber phones.
>>
>> On Wed, Feb 3, 2016 at 2:28 PM, Bob Hartwig <bobjectsfreeswitch at gmail.com
>> > wrote:
>>
>>> I have a client that needs to reliably detect if their outbound calls
>>> are answered by a human or voicemail system, so that they can take
>>> different actions based on that determination.
>>>
>>> I looked at the AVMD module documentation at
>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd , and it
>>> seems to indicate that this simply detects a beep, i.e. it does not use
>>> talking / silence heuristics into account to determine if the call is
>>> answered by a human or machine.
>>>
>>> Am I correct about that?  How is this intended to be used, do I assume
>>> that the call is answered by a human until this module sends
>>> a avmd%3A%3Abeep event to me?  How do others use this module or other
>>> techniques to determine human / machine answer to outbound calls with
>>> Freeswitch?
>>>
>>> Thanks!
>>> Bob
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 dre at imerchantsystems.com  Thu Feb  4 23:29:33 2016
From: dre at imerchantsystems.com (Londre)
Date: Thu, 4 Feb 2016 15:29:33 -0500
Subject: [Freeswitch-users] mod_xml_curl
Message-ID: <037a01d15f8a$c24e1e00$46ea5a00$@imerchantsystems.com>

Hi,

 

I have read the wiki several times and followed all the steps to the best of
my knowledge. I have gotten mod_xml_curl build and making requests. I set
xml_curl debug_on and I can see the request that freeswitch sends and I can
take a look at the file it generates in the /tmp/ directory. However the
file doesn't seem to be writing anything. I have my web service set up to
hand a POST request but it looks like freeswitch is sending GET requests
even though I set the method to POST in xml_curl.xml. Any ideal would could
be going wrong?

 

System Info:

Debian Jessie

Freeswitch 1.7 built from source. 

 

Xml_curl.xml:

 

  1 <configuration name="xml_curl.conf" description="cURL XML Gateway">

  2     <bindings>

  3         <binding name="directory">

  4             <param name="gateway-url"
value="http://{domain}/api/directory" bindings="directory|dialplan"/>

  5             <param name="method" value="POST"/>

  6             <param name"gateway-credentials" value"user:pass"/>

  7         </binding>

  8     </bindings>

  9 </configuration>

 

It also doesn't seem to be sending the variables for the
gateway-credentials. Its seems weird that its sending a GET request when the
default should be a POST.  Also it is a little confusing on the wiki some
things show a query string and other show header variables. I'd prefer to
use POST. 

 

Please offer some insight into what could be going wrong. 

 

Londre Blocker

Developer

iMerchant Systems Inc.

T:  844.727.1998

 

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From bote_radio at botecomm.com  Fri Feb  5 00:44:23 2016
From: bote_radio at botecomm.com (Bote Man)
Date: Thu, 4 Feb 2016 16:44:23 -0500
Subject: [Freeswitch-users] X-Auth-IP Variable?
In-Reply-To: <CAAKbpsWKvTMcx_HxW7gDaGMjTfk6Ve-+-NbavXoeKY0Cb5NOdg@mail.gmail.com>
References: <CAAKbpsW9AS2jH+3Y5tQpytnrdSGbCkk_V6YewQ1s=HYs0Yi=ww@mail.gmail.com>	<CAMbW9oJSoijcU6_CMDzhsJEV_nVjmZPTvXoNVXQF03CKUYtEWw@mail.gmail.com>	<CAAKbpsVWMHMMc2i9y0XhpaTBV2gOTyiG3fDg7MrwoGz7AY8GZQ@mail.gmail.com>	<003c01d15f73$d07a5190$716ef4b0$@botecomm.com>
	<CAAKbpsWKvTMcx_HxW7gDaGMjTfk6Ve-+-NbavXoeKY0Cb5NOdg@mail.gmail.com>
Message-ID: <007701d15f95$365ceb00$a316c100$@botecomm.com>

Here?s what works for me:

 

<action application="log" data="NOTICE ***** network_addr is ${network_addr} ***** "/>

 

This will display it in fs_cli at least.

 

I?m not familiar with the syntax of what you posted, unless that?s an API command being pumped into FS via ESL or something.

 

Bote

 

 

From: Tim King
Sent: Thursday, 04 February, 2016 12:52
Subject: Re: [Freeswitch-users] X-Auth-IP Variable?

 

I keep trying to log them but the log application is showing me the log statement and not the variable itself. I know I have used this successfully before, but even if I try logging the destination_number it does not log.

 

I have tried ${destination_number} and [${destination_number}] and my log output looks like this.

 

Action log(Log TEST ========== ${destination_number} or [${destination_number}]

 

On Thu, Feb 4, 2016 at 12:45 PM, Bote Man <bote_radio at botecomm.com> wrote:

I also recommend using the FreeSWITCH ?log? application to display what FS thinks those variables contain. 

 

---

Bote

FreeSWITCH Docs Janitor

http://freeswitch.org/confluence

 

 

From: Tim King
Sent: Thursday, 04 February, 2016 12:13
Subject: Re: [Freeswitch-users] X-Auth-IP Variable?

Thank you for the reply. I tried this for matching to the ACL but it is failing.

<condition field="${acl(${variable_sip_h_X-Auth-IP} opensips)}" expression="true" break="on-false"/>

 

On Thu, Feb 4, 2016 at 11:44 AM, Andrew Cassidy <andrew at cassidywebservices.co.uk> wrote:

${variable_sip_h_X-Auth-IP}

 

On 4 February 2016 at 16:19, Tim King <tim.compnetwork at gmail.com> wrote:

I am using proxy authentication in my setup and it is working. To do this I have created an acl

autoload_configs/acl.conf.xml

   <list name="opensips" default="deny">

                <node type="allow" cidr="1.2.3.4/32"/>

                <node type="allow" cidr="1.2.3.5/32"/>

                <node type="allow" cidr="1.2.3.6/32"/>

   </list>

 

sip_profiles/external.xml

<param name="auth-calls" value="true"/>

<param name="apply-proxy-acl" value="1.2.3.2/32"/>

<param name="apply-inbound-acl" value="opensips"/>

 

 

This is all working as desired. The problem is prior to adding the opensips I was using the network_addr variable in my dialplan.

 

        <extension name="tollfree_to_providerA">

                <condition field="${acl(${network_addr} opensips)}" expression="true" break="on-false"/>

                        <condition field="destination_number" expression="^(\+?1)?(8(00|44|55|66|77|88)[2-9]\d{6})$">

                                <action application = "set" data="dtmf_type=rfc2833"/>

                                <action application="set" data="accountcode=customer1123"/>

                                <action application="set" data="continue_on_fail=false"/>

                                <action application="set" data="hangup_after_bridge=true"/>

                                <action application="set" data="proxy_media=true"/>

                                <action application="bridge" data="sofia/external/$2 at 8.7.6.5:5060"/>

                </condition>

        </extension>

 

This of course no longer works because network_addr is always the address of my proxy server. How can I get the address from the X-authip into the dialplan?

 

 

 

-- 

Andrew Cassidy BSc (Hons) MBCS SSCA

Managing Director

  <http://www.cassidywebservices.co.uk/static/emailsig.png> 

 

 <mailto:info at cassidywebservices.co.uk> T 03300 100 960   <mailto:info at cassidywebservices.co.uk> F 03300 100 961

 <mailto:info at cassidywebservices.co.uk> E andrew at cassidywebservices.co.uk

 <mailto:info at cassidywebservices.co.uk> W www.cassidywebservices.co.uk


_________________________________________________________________________




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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at 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 bote_radio at botecomm.com  Fri Feb  5 00:47:29 2016
From: bote_radio at botecomm.com (Bote Man)
Date: Thu, 4 Feb 2016 16:47:29 -0500
Subject: [Freeswitch-users] fs_cli under Windows and cygwin, no prompt
In-Reply-To: <56B3C09A.60907@telefaks.de>
References: <56B3C09A.60907@telefaks.de>
Message-ID: <007c01d15f95$a588b220$f09a1660$@botecomm.com>

You can only connect to port 8021 locally because

  <param name="listen-ip" value="127.0.0.1"/>

Is set in autoload_configs/event_socket.conf.xml

If you set that to value to 0.0.0.0 the whole world can connect to port
8021, but the problem with that is that the whole world can connect. I
wouldn't change without careful consideration of the security impact.

I have never used ssh to connect to Windows machine, I didn't even know that
it is possible so that is a new one to me. I just remote desktop (mstsc.exe)
into the target machine and run the necessary commands and utilities
directly on the box.

Sorry I couldn't be more help.


---
Bote
FreeSWITCH Docs Janitor
http://freeswitch.org/confluence




> -----Original Message-----
> From: Peter Steinbach
> Sent: Thursday, 04 February, 2016 16:20
> Subject: [Freeswitch-users] fs_cli under Windows and cygwin, no prompt
> 
> I have inherited a Freeswitch installation on a Windows server, which I
> remotely manage with cygwin and ssh.
> 
> This worked more or less nicely so far. But today I cannot manage
> Freeswitch via ssh with fs_cli anymore. I restarted the hardware, but no
> change.
> I can run fs_cli.exe, and I see the Freeswitch messages, but I do
> __not__ get a prompt. When I try to enter characters, nothing happens.
> 
> On the machine itself, when I open a console and start cygwin,
> fs_cli.exe works as expected with prompt. But not via ssh. But
> fs_cli.exe -x "command" works via ssh, so login to Freeswitch should be
> fine in general.
> 
> What I understand is, that fs_cli connects to Freeswitch via port 8021.
> So when seeing FS messages, I expect that the connection is already
> fine, right?
> 
> I also tried to connect via telnet to port 8021. This works in the local
> cygwin shell (get an auth request), but not via remote ssh shell (telnet
> terminates without message).
> 
> I am puzzled. Anybody has a clue, why I do not have a prompt and where
> to search?
> 
> --
> With kind regards
> Peter Steinbach
> 
> Telefaks Services GmbH
> mailto:lists (att) telefaks.de
> Internet: www.telefaks.de
> 
> 
> __________________________________________________________
> _______________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Fri Feb  5 01:06:25 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 4 Feb 2016 17:06:25 -0500
Subject: [Freeswitch-users] Best practices for determining if originated
	call is answered by human or voicemail system?
In-Reply-To: <CAL7Tj2_oKsj8dAZXTZXURcEgRDqpHsbVfdQzBCosfKWLKK529Q@mail.gmail.com>
References: <CAAemr3yoTC5obmo_SoGcyO+AehMOedRyQHSK41_d02dZrq_teA@mail.gmail.com>
	<CAD-qRkjhTHdWBxVgm5baWSV+fLFuPyvzYW4JLf+WB_REUVRN5g@mail.gmail.com>
	<CAL7Tj2_JVak-y=x-CpGyfMnqMZTkjjDTzaT93Pm3X1h5VN0RxA@mail.gmail.com>
	<CAL7Tj2_oKsj8dAZXTZXURcEgRDqpHsbVfdQzBCosfKWLKK529Q@mail.gmail.com>
Message-ID: <CBEE4850-A8A5-4B1D-9BFF-2DEF5CEEA07E@jerris.com>

See the previous response about our commercial module available if you are looking for something that detects segments of speech/silence.  That is exactly what that module does.

> On Feb 4, 2016, at 2:42 PM, Piotr Gregor <peterg at sytelco.com> wrote:
> 
> Hi,
> "I looked at the AVMD module documentation at https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd <https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd> , and it seems to indicate that this simply detects a beep, i.e. it does not use talking / silence heuristics into account to determine if the call is answered by a human or machine. 
> Am I correct about that? "
> 
> Yes. The AVMD module works by calculating the estimate of the frequency and amplitude of signal using DESA-2 algorithm. It fires amd::beep event when the variance for the frequency estimate is below the threshold.
> The event is of form:
> 
> Event-Subclass: avmd%3A%3Abeep
> Event-Name: CUSTOM
> Core-UUID: 36a5b214-8d83-487e-8f8b-50af9a090486
> FreeSWITCH-Hostname: home
> FreeSWITCH-Switchname: home
> FreeSWITCH-IPv4: 128.11.35.8
> FreeSWITCH-IPv6: %3A%3A1
> Event-Date-Local: 2016-02-04%2016%3A56%3A55
> Event-Date-GMT: Thu,%2004%20Feb%202016%2016%3A56%3A55%20GMT
> Event-Date-Timestamp: 1454605015915799
> Event-Calling-File: mod_avmd.c
> Event-Calling-Function: avmd_process
> Event-Calling-Line-Number: 556
> Event-Sequence: 950
> Beep-Status: stop
> Unique-ID: 0a6f6df3-cc48-4b8a-93f3-b35ee1b8ef0a
> call-command: avmd
> 
> The AVMD doesn't take into consideration segments of speech and segments of silence. It simply calculates it's estimate of frequency and amplitude.
> 
> 
> "How is this intended to be used, do I assume that the call is answered by a human until this module sends a avmd%3A%3Abeep event to me?  How do others use this module or other techniques to determine human / machine answer to outbound calls with Freeswitch? "
> 
> It's use is to detect beep only.
> You can test how it works by creating extension that will play a tone of given frequency, e.g:
>     <extension name="10014">
>       <condition field="destination_number" expression="^10014$">
>         <action application="playback" data="tone_stream://L=3;%(500,6850,850)" />
>         <action application="hangup"/>
>       </condition>
>     </extension>
> This plays tone of 500Hz frequency and 6850 ms long, then silence by 850 ms.
> 
> Start avmd on the call with fs_cli:
> avmd 8c6a6bb1-ec64-4b2c-9e48-93a6a2598b64 start
> 
> and inspect events by regestering to avmd events with:
> events plain CUSTOM avmd::beep
> 
> Analysing the audio for a presence of the tone is not enough for answering machine detection. You should also analyse at least the length of speech/silence segments. You can do this by subscription to TALK/NOTALK events.
> Hope this helps.
> 
> cheers,
> Piotr
> 
> On 4 February 2016 at 17:50, Piotr Gregor <peterg at sytelco.com <mailto:peterg at sytelco.com>> wrote:
> Hi,
> "I looked at the AVMD module documentation at https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd <https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd> , and it seems to indicate that this simply detects a beep, i.e. it does not use talking / silence heuristics into account to determine if the call is answered by a human or machine. 
> Am I correct about that? "
> 
> Yes. The AVMD module works by calculating the estimate of the frequency using DESA-2 algorithm. It fires amd::beep event when the variance for that estimate is below threshold.
> The event is of form:
> 
> Event-Subclass: avmd%3A%3Abeep
> Event-Name: CUSTOM
> Core-UUID: 36a5b214-8d83-487e-8f8b-50af9a090486
> FreeSWITCH-Hostname: home
> FreeSWITCH-Switchname: home
> FreeSWITCH-IPv4: 128.11.35.8
> FreeSWITCH-IPv6: %3A%3A1
> Event-Date-Local: 2016-02-04%2016%3A56%3A55
> Event-Date-GMT: Thu,%2004%20Feb%202016%2016%3A56%3A55%20GMT
> Event-Date-Timestamp: 1454605015915799
> Event-Calling-File: mod_avmd.c
> Event-Calling-Function: avmd_process
> Event-Calling-Line-Number: 556
> Event-Sequence: 950
> Beep-Status: stop
> Unique-ID: 0a6f6df3-cc48-4b8a-93f3-b35ee1b8ef0a
> call-command: avmd
> 
> The AVMD doesn't take into consideration TALK/NOTALK events.
> 
> 
> "How is this intended to be used, do I assume that the call is answered by a human until this module sends a avmd%3A%3Abeep event to me?  How do others use this module or other techniques to determine human / machine answer to outbound calls with Freeswitch? "
> 
> You can test how it works by creating extension that will play a tone of given frequency, e.g:
>     <extension name="10014">
>       <condition field="destination_number" expression="^10014$">
>         <action application="playback" data="tone_stream://L=3;%(500,6850,850)" />
>         <action application="hangup"/>
>       </condition>
>     </extension>
> This plays tone of 500Hz frequency and 6850 ms long, then silence by 850 ms.
> 
> Start avmd on the call with fs_cli:
> avmd 8c6a6bb1-ec64-4b2c-9e48-93a6a2598b64 start
> 
> and inspect events by regestering to avmd events with:
> events plain CUSTOM avmd::beep
> 
> Analysing the audio for a presence of the tone is not enough for answering machine detection. You should also analyse at least the length of speech/silence segments.
> Hope this helps.
> 
> cheers,
> Piotr
> 
> 
> 
> 
> 
> 
> 
> On 3 February 2016 at 19:50, Christopher Rienzo <cmrienzo at gmail.com <mailto:cmrienzo at gmail.com>> wrote:
> I wouldn't use beep detection in a dialer application, but it could be useful in something like follow me to reduce the occurrence of voicemails being left on subscriber phones.
> 
> On Wed, Feb 3, 2016 at 2:28 PM, Bob Hartwig <bobjectsfreeswitch at gmail.com <mailto:bobjectsfreeswitch at gmail.com>> wrote:
> I have a client that needs to reliably detect if their outbound calls are answered by a human or voicemail system, so that they can take different actions based on that determination.
> 
> I looked at the AVMD module documentation at https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd <https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd> , and it seems to indicate that this simply detects a beep, i.e. it does not use talking / silence heuristics into account to determine if the call is answered by a human or machine.  
> 
> Am I correct about that?  How is this intended to be used, do I assume that the call is answered by a human until this module sends a avmd%3A%3Abeep event to me?  How do others use this module or other techniques to determine human / machine answer to outbound calls with Freeswitch? 
> 
> Thanks!
> Bob
> 
> 
> 
> _________________________________________________________________________
> 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 Feb  5 03:02:51 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Feb 2016 18:02:51 -0600
Subject: [Freeswitch-users] Time for the Annual FreeSWITCH Developers Summit
Message-ID: <CAKbxfG_STTgNPpJQ3g7paSeGkN5r0w3KYQASRru8-GC-aHF1Mw@mail.gmail.com>

This weekend marks the beginning of the 2016 FreeSWITCH summit where the
team meets up to work on ClueCon and code and the next release of
FreeSWITCH.

Now is your chance to virtually buy the guys a Beer or help fund part of
their meal!

paypal at freeswitch.org http://bit.ly/1L1Azn9






-- 
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 max at nysolutions.com  Fri Feb  5 06:02:22 2016
From: max at nysolutions.com (Moishe Grunstein)
Date: Fri, 5 Feb 2016 03:02:22 +0000
Subject: [Freeswitch-users] Time for the Annual FreeSWITCH Developers
 Summit
In-Reply-To: <CAKbxfG_STTgNPpJQ3g7paSeGkN5r0w3KYQASRru8-GC-aHF1Mw@mail.gmail.com>
References: <CAKbxfG_STTgNPpJQ3g7paSeGkN5r0w3KYQASRru8-GC-aHF1Mw@mail.gmail.com>
Message-ID: <e4dd57fc5f1241cdac791fdfe9a83b41@nysolutions.com>

Link not working.

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 Anthony Minessale
Sent: Thursday, February 4, 2016 7:03 PM
To: Freeswitch-users <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Time for the Annual FreeSWITCH Developers Summit


This weekend marks the beginning of the 2016 FreeSWITCH summit where the team meets up to work on ClueCon and code and the next release of FreeSWITCH.

Now is your chance to virtually buy the guys a Beer or help fund part of their meal!
paypal at freeswitch.org<mailto:paypal at freeswitch.org> http://bit.ly/1L1Azn9





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

? http://freeswitch.org/  ? http://cluecon.com/  ? http://twitter.com/FreeSWITCH
? irc.freenode.net<http://irc.freenode.net> #freeswitch ? http://freeswitch.org/g+
ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>  ? +19193869900

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From blake at cogents.io  Fri Feb  5 06:40:27 2016
From: blake at cogents.io (Blake Priddy)
Date: Thu, 4 Feb 2016 21:40:27 -0600
Subject: [Freeswitch-users] Fellow FreeSWITCHERS
Message-ID: <CA+Kn03mfajU_rtSbib39UPnL_SgBuCd2KbCDVyJzz7Kg6k_b7g@mail.gmail.com>

It's time for the freeswitch annual dev meeting. The freeswitch developers
we love and care for so much will be gathering to talk business, future
development, and just talking and networking! I am saying this because I
have found out that most of the devs are paying out of their pocket for
their dinner, hotel, and travel. This is not okay with me. Therefore I have
made a sizeable donation on the freeswitch website for the developers to
use for food, room, flight, etc. I honestly don't care what is done with my
donation, I just know it will be put to good use. I am encouraging all of
you out there to spare some funds for this team that has helped us make
money, and/or help others save money. Let's give this team the meal they
deserve! Thanks freeswitch devs for all you do and continue to do to help
others save money and be successful in their VoIP implementations. We
appreciate you so much!!

Support the team that built the dream!

PayPal for the devs: paypal at freeswitch.org

or visit the website and click the donate button! https://freeswitch.org/
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From anthony.minessale at gmail.com  Fri Feb  5 09:24:57 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Feb 2016 00:24:57 -0600
Subject: [Freeswitch-users] Time for the Annual FreeSWITCH Developers
	Summit
In-Reply-To: <e4dd57fc5f1241cdac791fdfe9a83b41@nysolutions.com>
References: <CAKbxfG_STTgNPpJQ3g7paSeGkN5r0w3KYQASRru8-GC-aHF1Mw@mail.gmail.com>
	<e4dd57fc5f1241cdac791fdfe9a83b41@nysolutions.com>
Message-ID: <CAKbxfG-qsKej5tEfybT_dH9N-Q2K72Eh2F1snedQtA=FuCb9Kg@mail.gmail.com>

Oops

http://bit.ly/20KH4DR

The paypal button on the site works as well or just paypal at freeswitch.org
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From andrew at cassidywebservices.co.uk  Fri Feb  5 12:28:26 2016
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Fri, 5 Feb 2016 09:28:26 +0000
Subject: [Freeswitch-users] X-Auth-IP Variable?
In-Reply-To: <007701d15f95$365ceb00$a316c100$@botecomm.com>
References: <CAAKbpsW9AS2jH+3Y5tQpytnrdSGbCkk_V6YewQ1s=HYs0Yi=ww@mail.gmail.com>
	<CAMbW9oJSoijcU6_CMDzhsJEV_nVjmZPTvXoNVXQF03CKUYtEWw@mail.gmail.com>
	<CAAKbpsVWMHMMc2i9y0XhpaTBV2gOTyiG3fDg7MrwoGz7AY8GZQ@mail.gmail.com>
	<003c01d15f73$d07a5190$716ef4b0$@botecomm.com>
	<CAAKbpsWKvTMcx_HxW7gDaGMjTfk6Ve-+-NbavXoeKY0Cb5NOdg@mail.gmail.com>
	<007701d15f95$365ceb00$a316c100$@botecomm.com>
Message-ID: <CAMbW9o+c37vt=BGVrxM11TvaH=PyEus2ASJdkgOHBkSWeJXA+w@mail.gmail.com>

The "info" application dumps all channel variables

<action application="info" />



On 4 February 2016 at 21:44, Bote Man <bote_radio at botecomm.com> wrote:

> Here?s what works for me:
>
>
>
> <action application="log" data="NOTICE ***** network_addr is
> ${network_addr} ***** "/>
>
>
>
> This will display it in fs_cli at least.
>
>
>
> I?m not familiar with the syntax of what you posted, unless that?s an API
> command being pumped into FS via ESL or something.
>
>
>
> Bote
>
>
>
>
>
> *From:* Tim King
> *Sent:* Thursday, 04 February, 2016 12:52
>
> *Subject:* Re: [Freeswitch-users] X-Auth-IP Variable?
>
>
>
> I keep trying to log them but the log application is showing me the log
> statement and not the variable itself. I know I have used this successfully
> before, but even if I try logging the destination_number it does not log.
>
>
>
> I have tried ${destination_number} and [${destination_number}] and my log
> output looks like this.
>
>
>
> Action log(Log TEST ========== ${destination_number} or
> [${destination_number}]
>
>
>
> On Thu, Feb 4, 2016 at 12:45 PM, Bote Man <bote_radio at botecomm.com> wrote:
>
> I also recommend using the FreeSWITCH ?log? application to display what FS
> thinks those variables contain.
>
>
>
> ---
>
> Bote
>
> FreeSWITCH Docs Janitor
>
> http://freeswitch.org/confluence
>
>
>
>
>
> *From:* Tim King
> *Sent:* Thursday, 04 February, 2016 12:13
> *Subject:* Re: [Freeswitch-users] X-Auth-IP Variable?
>
> Thank you for the reply. I tried this for matching to the ACL but it is
> failing.
>
> <condition field="${acl(${variable_sip_h_X-Auth-IP} opensips)}"
> expression="true" break="on-false"/>
>
>
>
> On Thu, Feb 4, 2016 at 11:44 AM, Andrew Cassidy <
> andrew at cassidywebservices.co.uk> wrote:
>
> ${variable_sip_h_X-Auth-IP}
>
>
>
> On 4 February 2016 at 16:19, Tim King <tim.compnetwork at gmail.com> wrote:
>
> I am using proxy authentication in my setup and it is working. To do this
> I have created an acl
>
> *autoload_configs/acl.conf.xml*
>
>    <list name="opensips" default="deny">
>
>                 <node type="allow" cidr="1.2.3.4/32"/>
>
>                 <node type="allow" cidr="1.2.3.5/32"/>
>
>                 <node type="allow" cidr="1.2.3.6/32"/>
>
>    </list>
>
>
>
> *sip_profiles/external.xml*
>
> <param name="auth-calls" value="true"/>
>
> <param name="apply-proxy-acl" value="1.2.3.2/32"/>
>
> <param name="apply-inbound-acl" value="opensips"/>
>
>
>
>
>
> This is all working as desired. The problem is prior to adding the
> opensips I was using the network_addr variable in my dialplan.
>
>
>
>         <extension name="tollfree_to_providerA">
>
>                 <condition field="${acl(${network_addr} opensips)}"
> expression="true" break="on-false"/>
>
>                         <condition field="destination_number"
> expression="^(\+?1)?(8(00|44|55|66|77|88)[2-9]\d{6})$">
>
>                                 <action application = "set"
> data="dtmf_type=rfc2833"/>
>
>                                 <action application="set"
> data="accountcode=customer1123"/>
>
>                                 <action application="set"
> data="continue_on_fail=false"/>
>
>                                 <action application="set"
> data="hangup_after_bridge=true"/>
>
>                                 <action application="set"
> data="proxy_media=true"/>
>
>                                 <action application="bridge"
> data="sofia/external/$2 at 8.7.6.5:5060"/>
>
>                 </condition>
>
>         </extension>
>
>
>
> This of course no longer works because network_addr is always the address
> of my proxy server. How can I get the address from the X-authip into the
> dialplan?
>
>
>
>
>
>
>
> --
>
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>
> Managing Director
>
>
>
> *T <info at cassidywebservices.co.uk> *03300 100 960  *F
> <info at cassidywebservices.co.uk> *03300 100 961
>
> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
>
> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>
>
> _________________________________________________________________________
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960  *F
<info at cassidywebservices.co.uk> *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From lists at telefaks.de  Fri Feb  5 12:30:44 2016
From: lists at telefaks.de (Peter Steinbach)
Date: Fri, 05 Feb 2016 10:30:44 +0100
Subject: [Freeswitch-users] fs_cli under Windows and cygwin, no prompt
In-Reply-To: <007c01d15f95$a588b220$f09a1660$@botecomm.com>
References: <56B3C09A.60907@telefaks.de>
	<007c01d15f95$a588b220$f09a1660$@botecomm.com>
Message-ID: <56B46BC4.1010607@telefaks.de>

Hello Bote,

thanks for your reply. Maybe my post was a bit misleading. In both cases
I have a shell opened on the machine and try from there running
fs_cli.exe. The difference is

  * 1st case: I open a local shell on the Windows machine (via RDP
    connection), run cygwin, get a unix shell and then start fs_cli. Works.
  * 2nd case: Form an external machine I ssh to the Windows machine
    (sshd server is running there with cygwin), have a unix shell and
    then start fs_cli. Does not work as described below.

So both cases are connecting to localhost:8021. And I am wondering now,
why this behaves differently.

Why Cygwin? For me, this is much easier, because I am used to Linux
tools (grep, tail, editors, mc, ...). And it's much better for
documentation purposes, when this all is text based and you have a large
terminal window.


Best regards
Peter

On 02/04/16 22:47, Bote Man wrote:
> You can only connect to port 8021 locally because
>
>   <param name="listen-ip" value="127.0.0.1"/>
>
> Is set in autoload_configs/event_socket.conf.xml
>
> If you set that to value to 0.0.0.0 the whole world can connect to port
> 8021, but the problem with that is that the whole world can connect. I
> wouldn't change without careful consideration of the security impact.
>
> I have never used ssh to connect to Windows machine, I didn't even know that
> it is possible so that is a new one to me. I just remote desktop (mstsc.exe)
> into the target machine and run the necessary commands and utilities
> directly on the box.
>
> Sorry I couldn't be more help.
>
>
> ---
> Bote
> FreeSWITCH Docs Janitor
> http://freeswitch.org/confluence
>
>
>
>
>> -----Original Message-----
>> From: Peter Steinbach
>> Sent: Thursday, 04 February, 2016 16:20
>> Subject: [Freeswitch-users] fs_cli under Windows and cygwin, no prompt
>>
>> I have inherited a Freeswitch installation on a Windows server, which I
>> remotely manage with cygwin and ssh.
>>
>> This worked more or less nicely so far. But today I cannot manage
>> Freeswitch via ssh with fs_cli anymore. I restarted the hardware, but no
>> change.
>> I can run fs_cli.exe, and I see the Freeswitch messages, but I do
>> __not__ get a prompt. When I try to enter characters, nothing happens.
>>
>> On the machine itself, when I open a console and start cygwin,
>> fs_cli.exe works as expected with prompt. But not via ssh. But
>> fs_cli.exe -x "command" works via ssh, so login to Freeswitch should be
>> fine in general.
>>
>> What I understand is, that fs_cli connects to Freeswitch via port 8021.
>> So when seeing FS messages, I expect that the connection is already
>> fine, right?
>>
>> I also tried to connect via telnet to port 8021. This works in the local
>> cygwin shell (get an auth request), but not via remote ssh shell (telnet
>> terminates without message).
>>
>> I am puzzled. Anybody has a clue, why I do not have a prompt and where
>> to search?
>>
>> --
>> With kind regards
>> Peter Steinbach
>>
>> Telefaks Services GmbH
>> mailto:lists (att) telefaks.de
>> Internet: www.telefaks.de
>>
>>
>> __________________________________________________________
>> _______________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de

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From steveayre at gmail.com  Fri Feb  5 15:24:34 2016
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 5 Feb 2016 12:24:34 +0000
Subject: [Freeswitch-users] X-Auth-IP Variable?
In-Reply-To: <CAMbW9oJSoijcU6_CMDzhsJEV_nVjmZPTvXoNVXQF03CKUYtEWw@mail.gmail.com>
References: <CAAKbpsW9AS2jH+3Y5tQpytnrdSGbCkk_V6YewQ1s=HYs0Yi=ww@mail.gmail.com>
	<CAMbW9oJSoijcU6_CMDzhsJEV_nVjmZPTvXoNVXQF03CKUYtEWw@mail.gmail.com>
Message-ID: <CAFiqYu=7LBmzDPLjfW14q0ayHdAxEHfYPni5KWqbCfug3iZ=mw@mail.gmail.com>

It'll be ${sip_h_X-Auth-IP} - the variable_ prefix added by the info app is
only to differentiate variables from fields.

On 4 February 2016 at 16:44, Andrew Cassidy <andrew at cassidywebservices.co.uk
> wrote:

> ${variable_sip_h_X-Auth-IP}
>
> On 4 February 2016 at 16:19, Tim King <tim.compnetwork at gmail.com> wrote:
>
>> I am using proxy authentication in my setup and it is working. To do this
>> I have created an acl
>> *autoload_configs/acl.conf.xml*
>>    <list name="opensips" default="deny">
>>                 <node type="allow" cidr="1.2.3.4/32"/>
>>                 <node type="allow" cidr="1.2.3.5/32"/>
>>                 <node type="allow" cidr="1.2.3.6/32"/>
>>    </list>
>>
>> *sip_profiles/external.xml*
>> <param name="auth-calls" value="true"/>
>> <param name="apply-proxy-acl" value="1.2.3.2/32"/>
>> <param name="apply-inbound-acl" value="opensips"/>
>>
>>
>> This is all working as desired. The problem is prior to adding the
>> opensips I was using the network_addr variable in my dialplan.
>>
>>         <extension name="tollfree_to_providerA">
>>                 <condition field="${acl(${network_addr} opensips)}"
>> expression="true" break="on-false"/>
>>                         <condition field="destination_number"
>> expression="^(\+?1)?(8(00|44|55|66|77|88)[2-9]\d{6})$">
>>                                 <action application = "set"
>> data="dtmf_type=rfc2833"/>
>>                                 <action application="set"
>> data="accountcode=customer1123"/>
>>                                 <action application="set"
>> data="continue_on_fail=false"/>
>>                                 <action application="set"
>> data="hangup_after_bridge=true"/>
>>                                 <action application="set"
>> data="proxy_media=true"/>
>>                                 <action application="bridge"
>> data="sofia/external/$2 at 8.7.6.5:5060"/>
>>                 </condition>
>>         </extension>
>>
>> This of course no longer works because network_addr is always the address
>> of my proxy server. How can I get the address from the X-authip into the
>> dialplan?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
>
>
> *T <info at cassidywebservices.co.uk> *03300 100 960  *F
> <info at cassidywebservices.co.uk> *03300 100 961
> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 italo at freeswitch.org  Fri Feb  5 21:44:27 2016
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Fri, 5 Feb 2016 15:44:27 -0300
Subject: [Freeswitch-users] Answered/abandoned calls mod_callcenter
In-Reply-To: <701581496dd4988b32ee3289cc2166df@mail2.boxtub.com>
References: <701581496dd4988b32ee3289cc2166df@mail2.boxtub.com>
Message-ID: <CABMX_w-2=si4EVhFvs+7OtickCj4eMaiu+zpPwgc4RMc4Snmyg@mail.gmail.com>

Hi Rodrigo,

We don't keep these counters in memory.

If you want to figure out right now how many calls were abandoned or
answered you need to parse the cdrs from these calls or listen to the ESL
events and keep a counter.

But, adding realtime counters to mod_callcenter shouldn't be difficult, a
PR would be awesome ;)


On Wed, Feb 3, 2016 at 9:35 PM, Rodrigo Ram?rez Norambuena <
decipher.hk at gmail.com> wrote:

> Hello everyone!,
>
> I'm testing a mod_callcenter for make a FreeSWITCH version of a open
> source software
> (https://github.com/roramirez/qpanel/tree/fs)
>
> Now i using ESL to send command and get information from the module. I
> looking for a way to know the
> data of answered/abandoned call from a queue and agents.
>
> Somebody can give a tips or a light?
>
> Regards,
> --
> Rodrigo Ram?rez Norambuena
> http://www.rodrigoramirez.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
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From italo at freeswitch.org  Fri Feb  5 21:51:11 2016
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Fri, 5 Feb 2016 15:51:11 -0300
Subject: [Freeswitch-users] mod_xml_curl
In-Reply-To: <037a01d15f8a$c24e1e00$46ea5a00$@imerchantsystems.com>
References: <037a01d15f8a$c24e1e00$46ea5a00$@imerchantsystems.com>
Message-ID: <CABMX_w-u6kovdmu7RGQvRQgzxwvoZjWgw54EVL=dEz5XgMtc+Q@mail.gmail.com>

Londre,

{domain} should be ${domain}

If mod_xml_curl is not respecting your configuration file then we have a
bug, can you please file a JIRA with your configs and log files?

On Thu, Feb 4, 2016 at 5:29 PM, Londre <dre at imerchantsystems.com> wrote:

> Hi,
>
>
>
> I have read the wiki several times and followed all the steps to the best
> of my knowledge. I have gotten mod_xml_curl build and making requests. I
> set xml_curl debug_on and I can see the request that freeswitch sends and I
> can take a look at the file it generates in the /tmp/ directory. However
> the file doesn?t seem to be writing anything. I have my web service set up
> to hand a POST request but it looks like freeswitch is sending GET requests
> even though I set the method to POST in xml_curl.xml. Any ideal would could
> be going wrong?
>
>
>
> System Info:
>
> Debian Jessie
>
> Freeswitch 1.7 built from source.
>
>
>
> Xml_curl.xml:
>
>
>
>   1 <configuration name="xml_curl.conf" description="cURL XML Gateway">
>
>   2     <bindings>
>
>   3         <binding name="directory">
>
>   4             <param name="gateway-url" value="http://{domain}/api/directory"
> bindings="directory|dialplan"/>
>
>   5             <param name="method" value="POST"/>
>
>   6             <param name"gateway-credentials" value"user:pass"/>
>
>   7         </binding>
>
>   8     </bindings>
>
>   9 </configuration>
>
>
>
> It also doesn?t seem to be sending the variables for the
> gateway-credentials. Its seems weird that its sending a GET request when
> the default should be a POST.  Also it is a little confusing on the wiki
> some things show a query string and other show header variables. I?d prefer
> to use POST.
>
>
>
> Please offer some insight into what could be going wrong.
>
>
>
> *Londre Blocker*
>
> Developer
>
> iMerchant Systems Inc.
>
> T:  844.727.1998
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From nneul at mst.edu  Sat Feb  6 00:45:28 2016
From: nneul at mst.edu (Nathan Neulinger)
Date: Fri, 5 Feb 2016 15:45:28 -0600
Subject: [Freeswitch-users] FYI in case anyone else was using the CallCap
	blocklist
Message-ID: <56B517F8.8070200@mst.edu>

It's been taken offline by the company, and they indicate they have no plans to bring the XML download back.

Is anyone here making any use of a similar data feed, or the FTC robocall complaint data or similar? Any recommendations?

-- Nathan

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


From peter at hartmanncomputer.com  Sat Feb  6 01:01:40 2016
From: peter at hartmanncomputer.com (Peter Hartmann)
Date: Fri, 5 Feb 2016 17:01:40 -0500
Subject: [Freeswitch-users] mod_esf with polycom
In-Reply-To: <003f01d15b95$07d0e380$1772aa80$@botecomm.com>
References: <CALXC2kcoroLZT47aBzhOumcGX1cMhV=g6fJ2+B3q=5Hx+Oa3SA@mail.gmail.com>
	<CAEJMVkCcovS8z_mFCKQTXXn=G6cm3oz4gyT5+hVUfU4qAhkOSQ@mail.gmail.com>
	<CAEJMVkDecUB-w8iGRy6G1rumNpsHcq+SjG=BXE6-zxr7AgoBPg@mail.gmail.com>
	<CALXC2kdq6Pyj4_Jjhh1VO=mMXb2e4hPp104YGn1DQCyyTq8Y0w@mail.gmail.com>
	<008001d1598c$ef0aa200$cd1fe600$@botecomm.com>
	<CALXC2kd1EYd+88esufdzKDuoWB_Cp1Bx3km8jfzQR370WGmkEQ@mail.gmail.com>
	<003f01d15b95$07d0e380$1772aa80$@botecomm.com>
Message-ID: <CALXC2kesAicqaBvq6OSG__Gr+7r6dcvUs=0i7_WZDtDnXfkf9g@mail.gmail.com>

> What do you mean by ?group??

It's a Polycom thing...check out the pdf Brian referenced.   Channel
is also a polycom thing.

http://support.polycom.com/global/documents/support/technical/products/voice/Audio_Packet_Format.pdf

I got it going.....it turns our that Paging Group 25 is channel 50 at
the packet level.

Thank you, Brian!




Peter Hartmann
Hartmann Computer Consulting
http://blog.hartmanncomputer.com
(212)203-8870

If I can't explain it to you in plain language, that means I don't
understand it.

On Sat, Jan 30, 2016 at 2:32 PM, Bote Man <bote_radio at botecomm.com> wrote:
>
> At this point I would try specifying all fields whether they are needed or not. Try
>
> <action application="esf_page_group" data="224.0.1.116 5001 6061 4"/>
>
> Where 4 is the TTL; I?m just guessing that maybe these packets must hop through a router or two.  The 6061 does not apply, but must be the third argument to get to the fourth argument.
>
>
>
> What do you mean by ?group?? The arguments specified on the Confluence page are the only ones that control the multicast page application. If Polycom needs additional processing then this app won?t work for you. I have never used Polycom so I can?t say, maybe others on here who have used Polycom could chime in.
>
>
>
> FYI, the multicast stream is sent out via UDP.
>
>
>
>
>
> ---
>
> Bote
>
>
>
> FreeSWITCH Docs Janitor
>
> http://freeswitch.org/confluence
>
>
>
>
>
>
>
>
>
>
>
>
>
> From: Peter Hartmann
> Sent: Friday, 29 January, 2016 14:41
>
>
> Subject: Re: [Freeswitch-users] mod_esf with polycom
>
>
>
> I just did a packet capture of a mod_esf call.  What I'm seeing is not a lot compared to a polycom page capture.  In the mod_esf cap, I just  a handful of packets of protocol IGMPv3.  Regardless of my config:
>
> <action application="esf_page_group" data="224.0.1.116 5001"/>
>
> They are being sent to the wrong destination address 224.0.0.22. And curiously, I see my destination address being used as the group.  (see attachment)
>
>
>
> I don't see this 224.0.0.22 address in the source for mod_esf at all....I wonder where that's coming from?
>
>
>
> In the polycom capture, there are no such IGMP packets, just UDP.   Although I do realize that it's supposed to do a few different broadcast types from different manufacturers all at once.
>
>
>
>
>
> Peter Hartmann
> Hartmann Computer Consulting
> http://blog.hartmanncomputer.com
> (212)203-8870
>
> If I can't explain it to you in plain language, that means I don't understand it.
>
>
>
>
>
> On Thu, Jan 28, 2016 at 12:30 AM, Bote Man <bote_radio at botecomm.com> wrote:
>
> Perhaps you are being led astray by the term ?multicast group? in the usage prototype. That is simply the multicast address, such as 239.1.1.1
>
>
>
> FIXED!
>
>
>
> IGMP = Internet Group Management Protocol which is where that came from.
>
>
>
> I have used mod_esf to send multicast streams to devices manufactured by one of my clients and it worked swimmingly with only the multicast address and port being the fields that mattered. Thanks to a change by anthm you can even add it to the conference bridge if you want both regular SIP calls and pages to be combined into one group announcement.
>
>
>
> With a phone there?s no telling what else they might require to light it up for a multicast announcement. Somebody was just asking a very similar question here last week for a different phone. If the phone requires some control signal or command to open up the speaker for a multicast announcement to follow, then I don?t think FS does that right now. It might be worth scouring Polycom documentation to see if they reveal any secrets.
>
>
>
> * * *
>
> I really wish the order of the arguments were
>
> Address, port, TTL, [other stuff]
>
> So that ?other stuff? could?ve been made optional. I have never gotten a clear definition of what that Linksys control port does, but my research revealed that it was likely carried over from an Asterisk compatibility feature.
>
>
>
>
>
> ---
>
> Bote
>
>
>
> FreeSWITCH Docs Janitor
>
> http://freeswitch.org/confluence
>
>
>
>
>
>
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Hartmann
> Sent: Wednesday, 27 January, 2016 22:27
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] mod_esf with polycom
>
>
>
> Thanks Brian.  Yes, I've tried it.  It looks promising though, since I can see some of  the default polycom settings defined in variables.  Nothing about group is mentioned...so maybe it's going to the default group?  I have a feeling I'll be doing a packet capture on a group-25 (emergency) broadcast to see what it does.
>
>
>
>
>
> Peter Hartmann
> Hartmann Computer Consulting
> http://blog.hartmanncomputer.com
> (212)203-8870
>
> If I can't explain it to you in plain language, that means I don't understand it.
>
>
>
> On Wed, Jan 27, 2016 at 10:03 PM, Brian West <brian at freeswitch.org> wrote:
>
> I'll review the source in the morning and verify if it's documented correctly.
>
>
>
> On Wednesday, January 27, 2016, Brian West <brian at freeswitch.org> wrote:
>
> Have you tried it?  It should just work if I recall correctly.
>
> On Wednesday, January 27, 2016, Peter Hartmann <peter at hartmanncomputer.com> wrote:
>
> Hey, how do you use mod_esf with Polycom?  The source suggests that
> it's supported, but the arguments in the documentation seem specific
> to Linksys.  From using the feature directly from a handset, I imagine
> we have to pass multicast-address, multicast-port and group-number.
> I've tried every combination and permutation with no luck.
>
> The reason I'm doing this is because if the emergency group number is
> used, the announcements will be at full volume regardless of the
> hands-free volume setting.
>
> And also I'm looking to do this in FS because I'm bridging in another
> PA system also.
>
>
> Thanks,
>
>
> Peter Hartmann
> Hartmann Computer Consulting
> http://blog.hartmanncomputer.com
> (212)203-8870
>
> If I can't explain it to you in plain language, that means I don't
> understand it.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
>
> Got Bugs? Report them here! | Reddit: /r/freeswitch
>
> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
> iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
>
>
>
>
>
> --
>
> Brian West
> brian at freeswitch.org
>
> Twitter: @FreeSWITCH , @briankwest
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> Got Bugs? Report them here! | Reddit: /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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Sat Feb  6 03:37:33 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Feb 2016 18:37:33 -0600
Subject: [Freeswitch-users] Fellow FreeSWITCHERS
In-Reply-To: <CA+Kn03mfajU_rtSbib39UPnL_SgBuCd2KbCDVyJzz7Kg6k_b7g@mail.gmail.com>
References: <CA+Kn03mfajU_rtSbib39UPnL_SgBuCd2KbCDVyJzz7Kg6k_b7g@mail.gmail.com>
Message-ID: <CAKbxfG-k6iDW9oW7ad3Ab-SwZcOeE8XhiMTdZJy_W_zf7zTxPg@mail.gmail.com>

Thanks so much Blake!


On Thu, Feb 4, 2016 at 9:40 PM, Blake Priddy <blake at cogents.io> wrote:

> It's time for the freeswitch annual dev meeting. The freeswitch developers
> we love and care for so much will be gathering to talk business, future
> development, and just talking and networking! I am saying this because I
> have found out that most of the devs are paying out of their pocket for
> their dinner, hotel, and travel. This is not okay with me. Therefore I have
> made a sizeable donation on the freeswitch website for the developers to
> use for food, room, flight, etc. I honestly don't care what is done with my
> donation, I just know it will be put to good use. I am encouraging all of
> you out there to spare some funds for this team that has helped us make
> money, and/or help others save money. Let's give this team the meal they
> deserve! Thanks freeswitch devs for all you do and continue to do to help
> others save money and be successful in their VoIP implementations. We
> appreciate you so much!!
>
> Support the team that built the dream!
>
> PayPal for the devs: paypal at freeswitch.org
>
> or visit the website and click the donate button! https://freeswitch.org/
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 deepikay at iiitd.ac.in  Sat Feb  6 17:24:02 2016
From: deepikay at iiitd.ac.in (Deepika Yadav)
Date: Sat, 6 Feb 2016 19:54:02 +0530
Subject: [Freeswitch-users] Suspicious Incoming Calls
Message-ID: <CAM99_YmfnTtEH6O1x=PQ378FUfFqLYO+p_40V0SayYqwyeUwQg@mail.gmail.com>

Hi,

I have microsip installed in my windows configured for one or two SIP
accounts for different Freeswitch servers. I am receiving a call from
2022 at myPublicIP repeatitively even if I disconnect from all the accounts,
these Freeswitch servers are hosted at cloud machine.

Is it a case of hacking the servers. What measures should I take to secure
both my servers and system.

Regards,
Deepika
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From brian at freeswitch.org  Sat Feb  6 17:28:33 2016
From: brian at freeswitch.org (Brian West)
Date: Sat, 6 Feb 2016 08:28:33 -0600
Subject: [Freeswitch-users] Fellow FreeSWITCHERS
In-Reply-To: <CA+Kn03mfajU_rtSbib39UPnL_SgBuCd2KbCDVyJzz7Kg6k_b7g@mail.gmail.com>
References: <CA+Kn03mfajU_rtSbib39UPnL_SgBuCd2KbCDVyJzz7Kg6k_b7g@mail.gmail.com>
Message-ID: <CAEJMVkD0EQW+kJ1JG3-ua5j7OtZ8MdqSW_Mt3v_ZD-jQ5N2pbg@mail.gmail.com>

Thank you!  You rock!

On Thursday, February 4, 2016, Blake Priddy <blake at cogents.io> wrote:

> It's time for the freeswitch annual dev meeting. The freeswitch developers
> we love and care for so much will be gathering to talk business, future
> development, and just talking and networking! I am saying this because I
> have found out that most of the devs are paying out of their pocket for
> their dinner, hotel, and travel. This is not okay with me. Therefore I have
> made a sizeable donation on the freeswitch website for the developers to
> use for food, room, flight, etc. I honestly don't care what is done with my
> donation, I just know it will be put to good use. I am encouraging all of
> you out there to spare some funds for this team that has helped us make
> money, and/or help others save money. Let's give this team the meal they
> deserve! Thanks freeswitch devs for all you do and continue to do to help
> others save money and be successful in their VoIP implementations. We
> appreciate you so much!!
>
> Support the team that built the dream!
>
> PayPal for the devs: paypal at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','paypal at freeswitch.org');>
>
> or visit the website and click the donate button! https://freeswitch.org/
>


-- 

*Brian West*
brian at freeswitch.org


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

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 jude19love at gmail.com  Sat Feb  6 18:50:23 2016
From: jude19love at gmail.com (Jude Mukundane)
Date: Sat, 6 Feb 2016 15:50:23 +0000
Subject: [Freeswitch-users] Suspicious Incoming Calls
In-Reply-To: <CAM99_YmfnTtEH6O1x=PQ378FUfFqLYO+p_40V0SayYqwyeUwQg@mail.gmail.com>
References: <CAM99_YmfnTtEH6O1x=PQ378FUfFqLYO+p_40V0SayYqwyeUwQg@mail.gmail.com>
Message-ID: <CAP=KgfaWH8e23UhYYhoQK4rRnhvEsOD3o=imzwJkbaycC5AT7g@mail.gmail.com>

Hello Deepika,

This is common for anyone running FS in the cloud with discoverable ports.
Lots of people just run scripts that crawl the internet in search of SIP
servers. After getting one they try bogus invites to try and see if they
can get calls through - if your server is forwarding to PSTN, you could end
up with a bill in thousands of dollars in minutest. In my case, I use a
simple IP Table in Ubuntu (more like an access control list) to define
allowed and non allowed IPs.

Can someone please elaborate on a measure that inolves config level
security because blocking out masses is not goot Internet Citizenry.

Jude

On Sat, Feb 6, 2016 at 2:24 PM, Deepika Yadav <deepikay at iiitd.ac.in> wrote:

> Hi,
>
> I have microsip installed in my windows configured for one or two SIP
> accounts for different Freeswitch servers. I am receiving a call from
> 2022 at myPublicIP repeatitively even if I disconnect from all the accounts,
> these Freeswitch servers are hosted at cloud machine.
>
> Is it a case of hacking the servers. What measures should I take to secure
> both my servers and system.
>
> 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 aqsyounas at gmail.com  Sat Feb  6 19:01:03 2016
From: aqsyounas at gmail.com (Aqs Younas)
Date: Sat, 6 Feb 2016 21:01:03 +0500
Subject: [Freeswitch-users] Forward call to a destination without changing
	RURI
Message-ID: <CAGVOjVXSEr5rb+3cHFyO=9rUO28rBPitxX926fiCqEh5K7QA5w@mail.gmail.com>

Hi,

I receive Invite from vendor with i  need to forward to a destination ip
set in special header X-Dest without changing the RURI.

When I try to bridge this invite

  <action application="bridge"
data="sofia/external/${destination}@${sip_h_X-Dest}" />

It changes the RURI. I need to preserve the RURI but set destination to
X-Dest header.

How can i forward call to that IP without changing the RURI.


Best Regard.
<https://www.avast.com/sig-email> This email has been sent from a
virus-free computer protected by Avast.
www.avast.com <https://www.avast.com/sig-email>
<#396301616_DDB4FAA8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
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From krice at freeswitch.org  Sat Feb  6 19:36:12 2016
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 6 Feb 2016 10:36:12 -0600
Subject: [Freeswitch-users] Suspicious Incoming Calls
In-Reply-To: <CAM99_YmfnTtEH6O1x=PQ378FUfFqLYO+p_40V0SayYqwyeUwQg@mail.gmail.com>
References: <CAM99_YmfnTtEH6O1x=PQ378FUfFqLYO+p_40V0SayYqwyeUwQg@mail.gmail.com>
Message-ID: <663601d160fc$7e239020$7a6ab060$@freeswitch.org>

http://youtu.be/oor-liSVL0o

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Deepika Yadav
Sent: Saturday, February 6, 2016 8:24 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Suspicious Incoming Calls

 

Hi, 

 

I have microsip installed in my windows configured for one or two SIP accounts for different Freeswitch servers. I am receiving a call from 2022 at myPublicIP repeatitively even if I disconnect from all the accounts, these Freeswitch servers are hosted at cloud machine.

 

Is it a case of hacking the servers. What measures should I take to secure both my servers and system.

 

Regards,

Deepika

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From s.safarov at gmail.com  Sat Feb  6 20:07:28 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 6 Feb 2016 20:07:28 +0300
Subject: [Freeswitch-users] Forward call to a destination without
	changing RURI
In-Reply-To: <CAGVOjVXSEr5rb+3cHFyO=9rUO28rBPitxX926fiCqEh5K7QA5w@mail.gmail.com>
References: <CAGVOjVXSEr5rb+3cHFyO=9rUO28rBPitxX926fiCqEh5K7QA5w@mail.gmail.com>
Message-ID: <CAHtxdDdHAw-83yEq3nbQxeRSb4qutBLWj0O+niREavBUOXPiNw@mail.gmail.com>

Try set sip_req_uri variable.

Sergey

On Sat, Feb 6, 2016 at 7:01 PM, Aqs Younas <aqsyounas at gmail.com> wrote:

> Hi,
>
> I receive Invite from vendor with i  need to forward to a destination ip
> set in special header X-Dest without changing the RURI.
>
> When I try to bridge this invite
>
>   <action application="bridge" data="sofia/external/${destination}@${sip_h_X-Dest}" />
>
> It changes the RURI. I need to preserve the RURI but set destination to
> X-Dest header.
>
> How can i forward call to that IP without changing the RURI.
>
>
> Best Regard.
> <https://www.avast.com/sig-email> This email has been sent from a
> virus-free computer protected by Avast.
> www.avast.com <https://www.avast.com/sig-email>
> <#-1084937732_396301616_DDB4FAA8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 ahabiba at gmail.com  Sat Feb  6 20:10:42 2016
From: ahabiba at gmail.com (Ahmed Habiba)
Date: Sat, 6 Feb 2016 20:10:42 +0300
Subject: [Freeswitch-users] Suspicious Incoming Calls
In-Reply-To: <mailman.1957.1454776624.678.freeswitch-users@lists.freeswitch.org>
References: <mailman.1957.1454776624.678.freeswitch-users@lists.freeswitch.org>
Message-ID: <CD155F84-618E-45AA-B550-F3FB002FE2FD@gmail.com>

Hello,

You have to use one of the below options:

Option1:
if you are not allowing another system to access you system without username and password i.e. you make your system as sip gateway for other trusted company, the  you can remove the file named ?external.xml? under /usr/local/freeswitch/conf/sip_profiles/, then either restart your instance or run ?reload mod_sofia? in fs_cli

Note be sure that you take a copy of external.xml before you remove it.

Option2:
add the below line in you external.xml profile mentioned above, this will not allow any external system to login expect if it has been allowed in you ACL list or it has a username/password this will make things little hard, then you may install fail2ban module.

all cases you need to restart your profiles.

Thanks,

Ahmed Habiba

> 
> 
> 
> From: Jude Mukundane <jude19love at gmail.com>
> Subject: Re: [Freeswitch-users] Suspicious Incoming Calls
> Date: February 6, 2016 at 6:50:23 PM GMT+3
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
> Hello Deepika, 
> 
> This is common for anyone running FS in the cloud with discoverable ports. Lots of people just run scripts that crawl the internet in search of SIP servers. After getting one they try bogus invites to try and see if they can get calls through - if your server is forwarding to PSTN, you could end up with a bill in thousands of dollars in minutest. In my case, I use a simple IP Table in Ubuntu (more like an access control list) to define allowed and non allowed IPs. 
> 
> Can someone please elaborate on a measure that inolves config level security because blocking out masses is not goot Internet Citizenry.
> 
> Jude
> 
> On Sat, Feb 6, 2016 at 2:24 PM, Deepika Yadav <deepikay at iiitd.ac.in <mailto:deepikay at iiitd.ac.in>> wrote:
> Hi, 
> 
> I have microsip installed in my windows configured for one or two SIP accounts for different Freeswitch servers. I am receiving a call from 2022 at myPublicIP repeatitively even if I disconnect from all the accounts, these Freeswitch servers are hosted at cloud machine.
> 
> Is it a case of hacking the servers. What measures should I take to secure both my servers and system.
> 
> Regards,
> Deepika
> 
> _________________________________________________________________________
> 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 aqsyounas at gmail.com  Sat Feb  6 20:52:42 2016
From: aqsyounas at gmail.com (Aqs Younas)
Date: Sat, 6 Feb 2016 22:52:42 +0500
Subject: [Freeswitch-users] Forward call to a destination without
	changing RURI
In-Reply-To: <CAHtxdDdHAw-83yEq3nbQxeRSb4qutBLWj0O+niREavBUOXPiNw@mail.gmail.com>
References: <CAGVOjVXSEr5rb+3cHFyO=9rUO28rBPitxX926fiCqEh5K7QA5w@mail.gmail.com>
	<CAHtxdDdHAw-83yEq3nbQxeRSb4qutBLWj0O+niREavBUOXPiNw@mail.gmail.com>
Message-ID: <CAGVOjVVu4NVJTG5JHfQW+GkSKeosuAmBvuwA-JKizsftyYit=Q@mail.gmail.com>

Thanks for your replay. Setting sip_req_uri variable was giving me internal
server error.

<anti-action application="bridge"
data="{sip_invite_req_rui=${sip_invite_req_rui}sofia/external/${
destination}@${sip_h_X-Dest}"/>

fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:40 sofia/external/14703999454 at 45.56.70.29:5070
Standard INIT
fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:48 (sofia/external/14703999454 at 45.56.70.29:5070)
State Change CS_INIT -> CS_ROUTING
fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:516 (sofia/external/14703999454 at 45.56.70.29:5070)
State INIT going to sleep
fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:473 (sofia/external/14703999454 at 45.56.70.29:5070)
Running State Change CS_ROUTING
fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG]
sofia.c:6760 Channel sofia/external/14703999454 at 45.56.70.29:5070 entering
state [terminated][900]
fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [NOTICE]
sofia.c:7779 Hangup sofia/external/14703999454 at 45.56.70.29:5070
[CS_ROUTING] [NORMAL_UNSPECIFIED]
fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:532 (sofia/external/14703999454 at 45.56.70.29:5070)
State ROUTING
fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG]
mod_sofia.c:141 sofia/external/14703999454 at 45.56.70.29:5070 SOFIA ROUTING
fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:532 (sofia/external/14703999454 at 45.56.70.29:5070)
State ROUTING going to sleep
fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:473 (sofia/external/14703999454 at 45.56.70.29:5070)
Running State Change CS_HANGUP
fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:739 (sofia/external/14703999454 at 45.56.70.29:5070)
Callstate Change DOWN -> HANGUP
fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:741 (sofia/external/14703999454 at 45.56.70.29:5070)
State HANGUP
fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG]
mod_sofia.c:431 Channel sofia/external/14703999454 at 45.56.70.29:5070 hanging
up, cause: NORMAL_UNSPECIFIED
fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:60 sofia/external/14703999454 at 45.56.70.29:5070
Standard HANGUP, cause: NORMAL_UNSPECIFIED
fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:741 (sofia/external/14703999454 at 45.56.70.29:5070)
State HANGUP going to sleep
fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:508 (sofia/external/14703999454 at 45.56.70.29:5070)
State Change CS_HANGUP -> CS_REPORTING
fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:473 (sofia/external/14703999454 at 45.56.70.29:5070)
Running State Change CS_REPORTING
fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:827 (sofia/external/14703999454 at 45.56.70.29:5070)
State REPORTING
fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:104 sofia/external/14703999454 at 45.56.70.29:5070
Standard REPORTING, cause: NORMAL_UNSPECIFIED
fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:827 (sofia/external/14703999454 at 45.56.70.29:5070)
State REPORTING going to sleep
9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [DEBUG]
switch_ivr_originate.c:3751 Originate Resulted in Error Cause: 31
[NORMAL_UNSPECIFIED]
fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:499 (sofia/external/14703999454 at 45.56.70.29:5070)
State Change CS_REPORTING -> CS_DESTROY
fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_session.c:1646 Session 50 (sofia/external/
14703999454 at 45.56.70.29:5070) Locked, Waiting on external entities
fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [NOTICE]
switch_core_session.c:1664 Session 50 (sofia/external/
14703999454 at 45.56.70.29:5070) Ended
    fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328
[NOTICE] switch_core_session.c:1668 Close Channel sofia/external/
14703999454 at 45.56.70.29:5070 [CS_DESTROY]
fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:630 (sofia/external/14703999454 at 45.56.70.29:5070)
Running State Change CS_DESTROY
9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [INFO]
mod_dptools.c:3379 Originate Failed.  Cause: NORMAL_UNSPECIFIED
9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [NOTICE]
switch_channel.c:4804 Hangup sofia/external/sipp at 104.237.140.13:5061
[CS_EXECUTE] [NORMAL_UNSPECIFIED]
fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:640 (sofia/external/14703999454 at 45.56.70.29:5070)
State DESTROY
fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG]
mod_sofia.c:341 sofia/external/14703999454 at 45.56.70.29:5070 SOFIA DESTROY
fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:111 sofia/external/14703999454 at 45.56.70.29:5070
Standard DESTROY
fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:640 (sofia/external/14703999454 at 45.56.70.29:5070)
State DESTROY going to sleep
9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_session.c:2796 sofia/external/sipp at 104.237.140.13:5061 skip
receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:539 (sofia/external/sipp at 104.237.140.13:5061)
State EXECUTE going to sleep
9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:473 (sofia/external/sipp at 104.237.140.13:5061)
Running State Change CS_HANGUP
9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:739 (sofia/external/sipp at 104.237.140.13:5061)
Callstate Change RINGING -> HANGUP
9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:741 (sofia/external/sipp at 104.237.140.13:5061)
State HANGUP
9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [DEBUG]
mod_sofia.c:425 sofia/external/sipp at 104.237.140.13:5061 Overriding SIP
cause 480 with 900 from the other leg
9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [DEBUG]
mod_sofia.c:431 Channel sofia/external/sipp at 104.237.140.13:5061 hanging up,
cause: NORMAL_UNSPECIFIED
9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [DEBUG]
mod_sofia.c:568 Responding to INVITE with: 900
9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:60 sofia/external/sipp at 104.237.140.13:5061
Standard HANGUP, cause: NORMAL_UNSPECIFIED
9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:741 (sofia/external/sipp at 104.237.140.13:5061)
State HANGUP going to sleep
9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:508 (sofia/external/sipp at 104.237.140.13:5061)
State Change CS_HANGUP -> CS_REPORTING
9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [DEBUG]
switch_core_state_machine.c:473 (sofia/external/sipp at 104.237.140.13:5061)
Running State Change CS_REPORTING
send 296 bytes to udp/[104.237.140.13]:5061 at 17:23:18.156113:


But got solution by using fs_path.

 <anti-action application="bridge"
data="sofia/external/${sip_req_uri};fs_path=sip:${sip_h_X-Dest}}"/>



On 6 February 2016 at 22:07, Sergey Safarov <s.safarov at gmail.com> wrote:

> Try set sip_req_uri variable.
>
> Sergey
>
> On Sat, Feb 6, 2016 at 7:01 PM, Aqs Younas <aqsyounas at gmail.com> wrote:
>
>> Hi,
>>
>> I receive Invite from vendor with i  need to forward to a destination ip
>> set in special header X-Dest without changing the RURI.
>>
>> When I try to bridge this invite
>>
>>   <action application="bridge" data="sofia/external/${destination}@${sip_h_X-Dest}" />
>>
>> It changes the RURI. I need to preserve the RURI but set destination to
>> X-Dest header.
>>
>> How can i forward call to that IP without changing the RURI.
>>
>>
>> Best Regard.
>> <https://www.avast.com/sig-email> This email has been sent from a
>> virus-free computer protected by Avast.
>> www.avast.com <https://www.avast.com/sig-email>
>> <#1808395836_-1084937732_396301616_DDB4FAA8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 koralu at gmail.com  Sat Feb  6 21:29:39 2016
From: koralu at gmail.com (Adrian Andrei)
Date: Sat, 6 Feb 2016 20:29:39 +0200
Subject: [Freeswitch-users] Run php script on execute_on_answer
Message-ID: <CABvcZyYh=DJMWimuLR=voa7B8PWnDRNwpEKU-qm4HKxUM1djVg@mail.gmail.com>

Hello,

I try to execute a php script after the call is answered but I can't figure
how. I need something like this.

<extension name="execOnAnswer">
<condition field="destination_number" expression="^07">
<action application="bridge" data="{execute_on_answer='run
http://localhost/answer.php?uuid=$uuid&bnr=$1'}sofia/external/$1 at 2.2.2.2"/>
</condition>
</extension>

So what I try to do is to pass the uuid and caller-destination-number to an
external source after the call is answered. Is any other approach?


Thank you
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From krice at freeswitch.org  Sat Feb  6 21:55:15 2016
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 6 Feb 2016 12:55:15 -0600
Subject: [Freeswitch-users] Run php script on execute_on_answer
In-Reply-To: <CABvcZyYh=DJMWimuLR=voa7B8PWnDRNwpEKU-qm4HKxUM1djVg@mail.gmail.com>
References: <CABvcZyYh=DJMWimuLR=voa7B8PWnDRNwpEKU-qm4HKxUM1djVg@mail.gmail.com>
Message-ID: <BA4EBDE7-87F6-4C2D-A48A-3AE1954FE601@freeswitch.org>

You cant run an http location you need to call it with curl

Sent from my iPhone

> On Feb 6, 2016, at 12:29 PM, Adrian Andrei <koralu at gmail.com> wrote:
> 
> Hello,
> 
> I try to execute a php script after the call is answered but I can't figure how. I need something like this.
> 
> <extension name="execOnAnswer">
> <condition field="destination_number" expression="^07">
> <action application="bridge" data="{execute_on_answer='run http://localhost/answer.php?uuid=$uuid&bnr=$1'}sofia/external/$1 at 2.2.2.2"/>
> </condition>
> </extension>
> 
> So what I try to do is to pass the uuid and caller-destination-number to an external source after the call is answered. Is any other approach?
> 
> 
> Thank you
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From koralu at gmail.com  Sat Feb  6 23:03:59 2016
From: koralu at gmail.com (Adrian Andrei)
Date: Sat, 6 Feb 2016 22:03:59 +0200
Subject: [Freeswitch-users] Run php script on execute_on_answer
In-Reply-To: <BA4EBDE7-87F6-4C2D-A48A-3AE1954FE601@freeswitch.org>
References: <CABvcZyYh=DJMWimuLR=voa7B8PWnDRNwpEKU-qm4HKxUM1djVg@mail.gmail.com>
	<BA4EBDE7-87F6-4C2D-A48A-3AE1954FE601@freeswitch.org>
Message-ID: <CABvcZyYgR3OtgRrdB6fvkpWFubY7mUrgAEXHowFCPYoCU6ye7Q@mail.gmail.com>

Ok. It works on a separate extension, but I can't figure out what is the
syntax for inline approach

<action data="{execute_on_answer='system curl
http://localhost/answer.php?uuid=$uuid&bnr=$1
<http://localhost/answer.php?uuid=$uuid&bnr=$1%27%7Dsofia/external/$1 at 2.2.2.2>
 inline=true'}sofia/external/$1 at 2.2.2.2"/>

This doesn't work.



On Sat, Feb 6, 2016 at 8:55 PM, Ken Rice <krice at freeswitch.org> wrote:

> You cant run an http location you need to call it with curl
>
> Sent from my iPhone
>
> On Feb 6, 2016, at 12:29 PM, Adrian Andrei <koralu at gmail.com> wrote:
>
> Hello,
>
> I try to execute a php script after the call is answered but I can't
> figure how. I need something like this.
>
> <extension name="execOnAnswer">
> <condition field="destination_number" expression="^07">
> <action application="bridge" data="{execute_on_answer='run
> http://localhost/answer.php?uuid=$uuid&bnr=$1'}sofia/external/$1 at 2.2.2.2
> "/>
> </condition>
> </extension>
>
> So what I try to do is to pass the uuid and caller-destination-number to
> an external source after the call is answered. Is any other approach?
>
>
> Thank you
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From denis.papes at zg.t-com.hr  Sat Feb  6 23:10:36 2016
From: denis.papes at zg.t-com.hr (=?UTF-8?Q?Denis_Pape=c5=a1?=)
Date: Sat, 6 Feb 2016 21:10:36 +0100
Subject: [Freeswitch-users] Run php script on execute_on_answer
In-Reply-To: <CABvcZyYgR3OtgRrdB6fvkpWFubY7mUrgAEXHowFCPYoCU6ye7Q@mail.gmail.com>
References: <CABvcZyYh=DJMWimuLR=voa7B8PWnDRNwpEKU-qm4HKxUM1djVg@mail.gmail.com>
	<BA4EBDE7-87F6-4C2D-A48A-3AE1954FE601@freeswitch.org>
	<CABvcZyYgR3OtgRrdB6fvkpWFubY7mUrgAEXHowFCPYoCU6ye7Q@mail.gmail.com>
Message-ID: <56B6533C.2090105@zg.t-com.hr>

You should load mod_curl and use syntax

<action data="{execute_on_answer='curl 
http://localhost/answer.php?uuid=$uuid&bnr=$1 
<http://localhost/answer.php?uuid=$uuid&bnr=$1%27%7Dsofia/external/$1 at 2.2.2.2>'}sofia/external/$1 at 2.2.2.2 
<mailto:1 at 2.2.2.2>"/>



On 02/06/2016 09:03 PM, Adrian Andrei wrote:
> Ok. It works on a separate extension, but I can't figure out what is 
> the syntax for inline approach
>
> <action data="{execute_on_answer='system curl 
> http://localhost/answer.php?uuid=$uuid&bnr=$1 
> <http://localhost/answer.php?uuid=$uuid&bnr=$1%27%7Dsofia/external/$1 at 2.2.2.2> inline=true'}sofia/external/$1 at 2.2.2.2 
> <mailto:1 at 2.2.2.2>"/>
>
> This doesn't work.
>
>
>
> On Sat, Feb 6, 2016 at 8:55 PM, Ken Rice <krice at freeswitch.org 
> <mailto:krice at freeswitch.org>> wrote:
>
>     You cant run an http location you need to call it with curl
>
>     Sent from my iPhone
>
>     On Feb 6, 2016, at 12:29 PM, Adrian Andrei <koralu at gmail.com
>     <mailto:koralu at gmail.com>> wrote:
>
>>     Hello,
>>
>>     I try to execute a php script after the call is answered but I
>>     can't figure how. I need something like this.
>>
>>     <extension name="execOnAnswer">
>>     <condition field="destination_number" expression="^07">
>>     <action application="bridge" data="{execute_on_answer='run
>>     http://localhost/answer.php?uuid=$uuid&bnr=$1'}sofia/external/$1 at 2.2.2.2
>>     <http://localhost/answer.php?uuid=$uuid&bnr=$1%27%7Dsofia/external/$1 at 2.2.2.2>"/>
>>     </condition>
>>     </extension>
>>
>>     So what I try to do is to pass the uuid and
>>     caller-destination-number to an external source after the call is
>>     answered. Is any other approach?
>>
>>
>>     Thank you
>>     _________________________________________________________________________
>>     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://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

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From koralu at gmail.com  Sat Feb  6 23:40:07 2016
From: koralu at gmail.com (Adrian Andrei)
Date: Sat, 6 Feb 2016 22:40:07 +0200
Subject: [Freeswitch-users] Run php script on execute_on_answer
In-Reply-To: <56B6533C.2090105@zg.t-com.hr>
References: <CABvcZyYh=DJMWimuLR=voa7B8PWnDRNwpEKU-qm4HKxUM1djVg@mail.gmail.com>
	<BA4EBDE7-87F6-4C2D-A48A-3AE1954FE601@freeswitch.org>
	<CABvcZyYgR3OtgRrdB6fvkpWFubY7mUrgAEXHowFCPYoCU6ye7Q@mail.gmail.com>
	<56B6533C.2090105@zg.t-com.hr>
Message-ID: <CABvcZyYuAB+d-t9E8DC+VrY=5V9xRkTQkO8cApXC1jWcGZiofQ@mail.gmail.com>

Mod_curl is loaded and in separate extension it's working fine. The problem
is with the following syntax:

I need something like this:

<action application="bridge" data="{execute_on_answer='curl
myLink?uuid=$uuid&bnr=$1'}sofia/external/$1at2.2.2.2"/>


On Sat, Feb 6, 2016 at 10:10 PM, Denis Pape? <denis.papes at zg.t-com.hr>
wrote:

> You should load mod_curl and use syntax
>
> <action data="{execute_on_answer='curl
> <http://localhost/answer.php?uuid=$uuid&bnr=$1%27%7Dsofia/external/$1 at 2.2.2.2>
> http://localhost/answer.php?uuid=$uuid&bnr=$1'}sofia/external/$
> <1 at 2.2.2.2>1 at 2.2.2.2"/>
>
>
>
> On 02/06/2016 09:03 PM, Adrian Andrei wrote:
>
> Ok. It works on a separate extension, but I can't figure out what is the
> syntax for inline approach
>
> <action data="{execute_on_answer='system curl
> <http://localhost/answer.php?uuid=$uuid&bnr=$1%27%7Dsofia/external/$1 at 2.2.2.2>
> http://localhost/answer.php?uuid=$uuid&bnr=$1
>  inline=true'}sofia/external/$ <1 at 2.2.2.2>1 at 2.2.2.2"/>
>
> This doesn't work.
>
>
>
> On Sat, Feb 6, 2016 at 8:55 PM, Ken Rice <krice at freeswitch.org> wrote:
>
>> You cant run an http location you need to call it with curl
>>
>> Sent from my iPhone
>>
>> On Feb 6, 2016, at 12:29 PM, Adrian Andrei < <koralu at gmail.com>
>> koralu at gmail.com> wrote:
>>
>> Hello,
>>
>> I try to execute a php script after the call is answered but I can't
>> figure how. I need something like this.
>>
>> <extension name="execOnAnswer">
>> <condition field="destination_number" expression="^07">
>> <action application="bridge" data="{execute_on_answer='run
>> <http://localhost/answer.php?uuid=$uuid&bnr=$1%27%7Dsofia/external/$1 at 2.2.2.2>
>> http://localhost/answer.php?uuid=$uuid&bnr=$1'}sofia/external/$1 at 2.2.2.2
>> "/>
>> </condition>
>> </extension>
>>
>> So what I try to do is to pass the uuid and caller-destination-number to
>> an external source after the call is answered. Is any other approach?
>>
>>
>> Thank you
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From denis.papes at zg.t-com.hr  Sat Feb  6 23:45:41 2016
From: denis.papes at zg.t-com.hr (=?UTF-8?Q?Denis_Pape=c5=a1?=)
Date: Sat, 6 Feb 2016 21:45:41 +0100
Subject: [Freeswitch-users] Run php script on execute_on_answer
In-Reply-To: <CABvcZyYuAB+d-t9E8DC+VrY=5V9xRkTQkO8cApXC1jWcGZiofQ@mail.gmail.com>
References: <CABvcZyYh=DJMWimuLR=voa7B8PWnDRNwpEKU-qm4HKxUM1djVg@mail.gmail.com>
	<BA4EBDE7-87F6-4C2D-A48A-3AE1954FE601@freeswitch.org>
	<CABvcZyYgR3OtgRrdB6fvkpWFubY7mUrgAEXHowFCPYoCU6ye7Q@mail.gmail.com>
	<56B6533C.2090105@zg.t-com.hr>
	<CABvcZyYuAB+d-t9E8DC+VrY=5V9xRkTQkO8cApXC1jWcGZiofQ@mail.gmail.com>
Message-ID: <56B65B75.9050402@zg.t-com.hr>

Sorry, I just copy/pasted your previous line without looking. It should be

<action application="bridge" data="{execute_on_answer=curl 
http://myLink?uuid=$uuid&bnr=$1}sofia/external/$1 at 2.2.2.2"/>


On 02/06/2016 09:40 PM, Adrian Andrei wrote:
> Mod_curl is loaded and in separate extension it's working fine. The 
> problem is with the following syntax:
>
> I need something like this:
>
> <action application="bridge" data="{execute_on_answer='curl 
> myLink?uuid=$uuid&bnr=$1'}sofia/external/$1at2.2.2.2"/>
>
>
> On Sat, Feb 6, 2016 at 10:10 PM, Denis Pape? <denis.papes at zg.t-com.hr 
> <mailto:denis.papes at zg.t-com.hr>> wrote:
>
>     You should load mod_curl and use syntax
>
>     <action data="{execute_on_answer='curl
>     http://localhost/answer.php?uuid=$uuid&bnr=$1'}sofia/external/$1 at 2.2.2.2
>     <mailto:1 at 2.2.2.2>"/>
>
>
>
>     On 02/06/2016 09:03 PM, Adrian Andrei wrote:
>>     Ok. It works on a separate extension, but I can't figure out what
>>     is the syntax for inline approach
>>
>>     <action data="{execute_on_answer='system curl
>>     http://localhost/answer.php?uuid=$uuid&bnr=$1 inline=true'}sofia/external/$1 at 2.2.2.2
>>     <mailto:1 at 2.2.2.2>"/>
>>
>>     This doesn't work.
>>
>>
>>
>>     On Sat, Feb 6, 2016 at 8:55 PM, Ken Rice <krice at freeswitch.org
>>     <mailto:krice at freeswitch.org>> wrote:
>>
>>         You cant run an http location you need to call it with curl
>>
>>         Sent from my iPhone
>>
>>         On Feb 6, 2016, at 12:29 PM, Adrian Andrei <koralu at gmail.com
>>         <mailto:koralu at gmail.com>> wrote:
>>
>>>         Hello,
>>>
>>>         I try to execute a php script after the call is answered but
>>>         I can't figure how. I need something like this.
>>>
>>>         <extension name="execOnAnswer">
>>>         <condition field="destination_number" expression="^07">
>>>         <action application="bridge" data="{execute_on_answer='run
>>>         http://localhost/answer.php?uuid=$uuid&bnr=$1'}sofia/external/$1 at 2.2.2.2
>>>         <mailto:1 at 2.2.2.2>"/>
>>>         </condition>
>>>         </extension>
>>>
>>>         So what I try to do is to pass the uuid and
>>>         caller-destination-number to an external source after the
>>>         call is answered. Is any other approach?
>>>
>>>
>>>         Thank you
>>>         _________________________________________________________________________
>>>         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://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 <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 krice at freeswitch.org  Sat Feb  6 23:56:28 2016
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 6 Feb 2016 14:56:28 -0600
Subject: [Freeswitch-users] Run php script on execute_on_answer
In-Reply-To: <CABvcZyYgR3OtgRrdB6fvkpWFubY7mUrgAEXHowFCPYoCU6ye7Q@mail.gmail.com>
References: <CABvcZyYh=DJMWimuLR=voa7B8PWnDRNwpEKU-qm4HKxUM1djVg@mail.gmail.com>
	<BA4EBDE7-87F6-4C2D-A48A-3AE1954FE601@freeswitch.org>
	<CABvcZyYgR3OtgRrdB6fvkpWFubY7mUrgAEXHowFCPYoCU6ye7Q@mail.gmail.com>
Message-ID: <5DFC87DE-7046-4EAF-A9A6-73506BBD9594@freeswitch.org>

Why call system there? See mod curl thats what its for

Sent from my iPhone

> On Feb 6, 2016, at 2:03 PM, Adrian Andrei <koralu at gmail.com> wrote:
> 
> Ok. It works on a separate extension, but I can't figure out what is the syntax for inline approach
> 
> <action data="{execute_on_answer='system curl http://localhost/answer.php?uuid=$uuid&bnr=$1 inline=true'}sofia/external/$1 at 2.2.2.2"/>
> 
> This doesn't work.
> 
> 
> 
>> On Sat, Feb 6, 2016 at 8:55 PM, Ken Rice <krice at freeswitch.org> wrote:
>> You cant run an http location you need to call it with curl
>> 
>> Sent from my iPhone
>> 
>>> On Feb 6, 2016, at 12:29 PM, Adrian Andrei <koralu at gmail.com> wrote:
>>> 
>>> Hello,
>>> 
>>> I try to execute a php script after the call is answered but I can't figure how. I need something like this.
>>> 
>>> <extension name="execOnAnswer">
>>> <condition field="destination_number" expression="^07">
>>> <action application="bridge" data="{execute_on_answer='run http://localhost/answer.php?uuid=$uuid&bnr=$1'}sofia/external/$1 at 2.2.2.2"/>
>>> </condition>
>>> </extension>
>>> 
>>> So what I try to do is to pass the uuid and caller-destination-number to an external source after the call is answered. Is any other approach?
>>> 
>>> 
>>> Thank you
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services: 
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From anton.vojlenko at gmail.com  Sun Feb  7 00:38:34 2016
From: anton.vojlenko at gmail.com (Anton)
Date: Sat, 6 Feb 2016 23:38:34 +0200
Subject: [Freeswitch-users] WebSocket behind NGINX
In-Reply-To: <56AD0CE7.6000607@gmail.com>
References: <56AD0CE7.6000607@gmail.com>
Message-ID: <56B667DA.4010505@gmail.com>

Hi,

Sorry for not answering for a long time.

Dan, thank you, your recommendation really helped me.

So in order to proxy websocket request you need:
1. Proxy websocket requests in this way WSS -> (NGINX) -> FS WSS or WS 
-> (NGINX) -> FS WS
2. Modify local-network-acl
3. Modify apply-candidate-acl if you would like to drop more rtp candidates

PS: I highly recommend to watch this video about NAT issues and ACL 
configuration:
https://www.youtube.com/watch?v=_WSx-T6TriI

BR,
Anton Voylenko

On 01/30/2016 09:20 PM, Anton wrote:
> Hello All,
>
> I have to proxy all websocket requests though a nginx server. Right 
> now I am using next configuration:
>
> map $http_upgrade $connection_upgrade {
>     default upgrade;
>     ''      close;
> }
>
> server {
>     listen 443;
>     server_name  wss.somedomain.com.ua;
>
>     ssl on;
>     ssl_certificate      /etc/nginx/cert.pem;
>     ssl_certificate_key  /etc/nginx/private.key;
>
>     location / {
>         proxy_pass http://127.0.0.1:5066;
>         proxy_http_version 1.1;
>         proxy_set_header Upgrade $http_upgrade;
>         proxy_set_header Connection $connection_upgrade;
>         proxy_read_timeout 86400s;
>    }
>
>     access_log /var/log/nginx/wss_access;
>     error_log /var/log/nginx/wss_error debug;
> }
>
> I dumped traffic from nginx and found out that "switching protocol" 
> phrase was successful but INVITE message from my browser in pending 
> state.
> Maybe FreeSWITCH wants real IP not loopback? Who have faced with 
> similar problem?
>
> BR,
> Anton



From william.suffill at gmail.com  Sun Feb  7 01:02:40 2016
From: william.suffill at gmail.com (William Suffill)
Date: Sat, 6 Feb 2016 17:02:40 -0500
Subject: [Freeswitch-users] FYI in case anyone else was using the
	CallCap blocklist
In-Reply-To: <56B517F8.8070200@mst.edu>
References: <56B517F8.8070200@mst.edu>
Message-ID: <CAF-ZooSBqRZmSJVZh6LuZo0DHGwnve+ECRatbfgZBt2mUgOFwQ@mail.gmail.com>

I'm also curious to what the community uses for this.

I been using TrueSpam  scores from TrueCNAM and it's be ok on my small
personal usage [Free Tier]. I also have been manually blocking callerids so
repeated calls get dropped automatically.

On Fri, Feb 5, 2016 at 4:45 PM, Nathan Neulinger <nneul at mst.edu> wrote:

> It's been taken offline by the company, and they indicate they have no
> plans to bring the XML download back.
>
> Is anyone here making any use of a similar data feed, or the FTC robocall
> complaint data or similar? Any recommendations?
>
> -- Nathan
>
> ------------------------------------------------------------
> Nathan Neulinger                       nneul at mst.edu
> Missouri S&T Information Technology    (573) 612-1412
> System Administrator - Architect
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gregor at infomedia.si  Sat Feb  6 21:36:39 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Sat, 06 Feb 2016 18:36:39 +0000
Subject: [Freeswitch-users] Run php script on execute_on_answer
In-Reply-To: <CABvcZyYh=DJMWimuLR=voa7B8PWnDRNwpEKU-qm4HKxUM1djVg@mail.gmail.com>
References: <CABvcZyYh=DJMWimuLR=voa7B8PWnDRNwpEKU-qm4HKxUM1djVg@mail.gmail.com>
Message-ID: <CAC7ttFGy0R609P+Kfe-FRt+5sGu1V3hDHN9kVYEPwQrkOU8ntA@mail.gmail.com>

Try to use https://wiki.freeswitch.org/wiki/Mod_curl

And execute it as api on answer.

On Sat, Feb 6, 2016, 19:31 Adrian Andrei <koralu at gmail.com> wrote:

> Hello,
>
> I try to execute a php script after the call is answered but I can't
> figure how. I need something like this.
>
> <extension name="execOnAnswer">
> <condition field="destination_number" expression="^07">
> <action application="bridge" data="{execute_on_answer='run
> http://localhost/answer.php?uuid=$uuid&bnr=$1'}sofia/external/$1 at 2.2.2.2
> "/>
> </condition>
> </extension>
>
> So what I try to do is to pass the uuid and caller-destination-number to
> an external source after the call is answered. Is any other approach?
>
>
> Thank you
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
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 gregor at infomedia.si  Sun Feb  7 08:51:08 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Sun, 7 Feb 2016 06:51:08 +0100
Subject: [Freeswitch-users] TCP registrations
In-Reply-To: <CAEJMVkB9vt4xGPo3Eeqke2C_MRJY9nJzqoURetUKg3ArP+sTLQ@mail.gmail.com>
References: <loom.20160201T233412-829@post.gmane.org>
	<001b01d15d51$455900d0$d00b0270$@botecomm.com>
	<CAC7ttFGGnvRDPHkCSDjwt2Ob0HXvBZv+_FXHCwXj8GFHHCuWHw@mail.gmail.com>
	<CAHtxdDeBNmV0KVxw-nL-BsX-c-q1UdRmxpFXoPuo6sjo9f0L+A@mail.gmail.com>
	<56B0ACD2.9070800@wirelessmundi.com>
	<CAEJMVkB9vt4xGPo3Eeqke2C_MRJY9nJzqoURetUKg3ArP+sTLQ@mail.gmail.com>
Message-ID: <CAC7ttFGPG0rwAKRHq4J3_VpnWO-C4bpX0Y+SR-_ULjQB4AUpsA@mail.gmail.com>

Everything works as expected :-) ALG on our router caused this problem. It
looks that ALG blocked outgoing SIP packages via TCP.

Thank you for your time.

2016-02-02 18:07 GMT+01:00 Brian West <brian at freeswitch.org>:

> I'm going to guess your device probably fails to send the transport=tcp on
> the contact there for it probably registers over TCP but we contact it back
> over UDP?  Can you confirm?
>
> On Tue, Feb 2, 2016 at 7:19 AM, Antonio Silva <asilva at wirelessmundi.com>
> wrote:
>
>> The parameter is "bind-params"
>>
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/Sofia+Configuration+Files
>>
>> by default fs should bind to tcp and udp but if you want only tcp just
>> set for the profile:
>>
>> <param name="bind-params" value="transport=tcp"/>
>>
>>
>>
>>
>> On 02/02/2016 02:01 PM, Sergey Safarov wrote:
>>
>> FS is defailt support UDP, TCP tansport. If you enable TLS in vars.xml
>> also TLS transport is will be enabled.
>> To check what is type of socket is open on server please use
>> netstat -an --inet | grep -P "5060|5061|5080"
>>
>> Example
>> [admin at node1.sbc ~]$ netstat -an --inet | grep -P "5060|5061|5080"
>> tcp        0      0 217.12.247.214:5060     0.0.0.0:*
>> LISTEN
>> tcp        0      0 10.21.7.30:5060
>> <http://t.sidekickopen35.com/e1t/c/5/f18dQhb0S7lC8dDMPbW2n0x6l2B9nMJN7t5XZsfmT10W5wLWvl4WYKY2W3MhGXn56dLgFf1Tdg-W02?t=http%3A%2F%2F10.21.7.30%3A5060%2F&si=5036397111279616&pi=d7b8e505-fa3b-49a0-bea1-31d998a61506>
>>         0.0.0.0:*               LISTEN
>> tcp        0      0 217.12.247.214:5061     0.0.0.0:*
>> LISTEN
>> tcp        0      0 217.12.247.214:5080     0.0.0.0:*
>> LISTEN
>> udp        0      0 217.12.247.214:5060     0.0.0.0:*
>>
>> udp        0      0 10.21.7.30:5060
>> <http://t.sidekickopen35.com/e1t/c/5/f18dQhb0S7lC8dDMPbW2n0x6l2B9nMJN7t5XZsfmT10W5wLWvl4WYKY2W3MhGXn56dLgFf1Tdg-W02?t=http%3A%2F%2F10.21.7.30%3A5060%2F&si=5036397111279616&pi=d7b8e505-fa3b-49a0-bea1-31d998a61506>
>>         0.0.0.0:*
>> udp        0      0 217.12.247.214:5080     0.0.0.0:*
>>
>> On Tue, Feb 2, 2016 at 11:50 AM, Gregor Nanger <gregor at infomedia.si>
>> wrote:
>>
>>> Yes, I also think so, but cannot find explicitly documented. So please,
>>> if anyone know exactly which command is, please help.
>>>
>>> 2016-02-02 1:32 GMT+01:00 Bote Man < <bote_radio at botecomm.com>
>>> bote_radio at botecomm.com>:
>>>
>>>> FreeSWITCH uses UDP by default for SIP signaling. You can change this
>>>> in the
>>>> SIP_profile I believe.
>>>>
>>>>
>>>> ---
>>>> Bote
>>>>
>>>> FreeSWITCH Docs Janitor
>>>> http://freeswitch.org/confluence
>>>>
>>>>
>>>>
>>>>
>>>> > -----Original Message-----
>>>> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>>> freeswitch-
>>>> > users-bounces at lists.freeswitch.org] On Behalf Of Gregor
>>>> > Sent: Monday, 01 February, 2016 17:37
>>>> > To: freeswitch-users at lists.freeswitch.org
>>>> > Subject: [Freeswitch-users] TCP registrations
>>>> >
>>>> > I think I am missing something.
>>>> >
>>>> > I would like to configure freeswitch that listens on TCP port for
>>>> client
>>>> > registrations (internal profile). As I read, freeswitch should do
>>>> this by
>>>> > default. But freeswitch responses only on UDP protocol. Is there a
>>>> conf
>>>> > setting for specify also tcp for registrations.
>>>> >
>>>> >
>>>> > __________________________________________________________
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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.orghttp://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>> --
>>
>> Saludos / Regards / Cumprimentos,
>> Ant?nio silva
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> Got Bugs? Report them here
> <http://t.sidekickopen35.com/e1t/c/5/f18dQhb0S7lC8dDMPbW2n0x6l2B9nMJN7t5XZsfmT10W5wLWvl4WYKY2W3MhGXn56dLgFf1Tdg-W02?t=https%3A%2F%2Ffreeswitch.org%2Fjira&si=5036397111279616&pi=d7b8e505-fa3b-49a0-bea1-31d998a61506>!
> | Reddit: /r/freeswitch
> <http://t.sidekickopen35.com/e1t/c/5/f18dQhb0S7lC8dDMPbW2n0x6l2B9nMJN7t5XZsfmT10W5wLWvl4WYKY2W3MhGXn56dLgFf1Tdg-W02?t=https%3A%2F%2Fwww.reddit.com%2Fr%2Ffreeswitch&si=5036397111279616&pi=d7b8e505-fa3b-49a0-bea1-31d998a61506>
>
> *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
>



-- 
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 vbvbrj at gmail.com  Sun Feb  7 17:59:42 2016
From: vbvbrj at gmail.com (Mimiko)
Date: Sun, 7 Feb 2016 16:59:42 +0200
Subject: [Freeswitch-users] hook to event or startup script which will hook
	to events?
Message-ID: <56B75BDE.6020604@gmail.com>

Hello.

I'm running FS without problems for 3 years. I installed back then FS 
version 1.3.13b cad607d72e and it is running till now. I know it is old, 
but its stable.

Back then I've used a start-up script in lua to catch callcenter's 
events and populate DB. Script is started and hooked to callcenter 
events. Now I want to add a event parse script for RECORD_STOP event. 
I've seen in lua.conf that I can use

<hook event="RECORD_STOP" script="script.lua"/>

to autohook and execute script for every RECORD_STOP event. For testing 
I've write a small script to just log info:

scriptname=argv[0]
freeswitch.consoleLog("notice", scriptname.." Starting...\n")
isdebug=false
min_rec_secs=0 -- In seconds.
api = freeswitch.API()
recordings_dir = api:execute("global_getvar", "recordings_dir")
databasename = api:execute("global_getvar", "freeswitch_data_db")
databaseuser = api:execute("global_getvar", "freeswitch_databaseuser")
databasepass = api:execute("global_getvar", "freeswitch_databasepass")
function printlog (a,...)
	-- First paramter - line number
	-- Second parameter - lvl=debug,info,notice,warning,err,crit,alert
	-- Third parameter - the message
	-- Calling: printlog([line number,][log lvl,]message)
	local arg={...}
	line=0
	lvl="debug"
	msg=a
	if (type(a)=='number' and a>0) then line=a else line=0 end
	maybe_lvl=string.lower(a)
	if (maybe_lvl=="debug" or maybe_lvl=="info" or maybe_lvl=="notice" or 
maybe_lvl=="warning" or maybe_lvl=="err" or maybe_lvl=="crit" or 
maybe_lvl=="alert") then
		lvl=maybe_lvl
	elseif (#arg>=1) then
		maybe_lvl=string.lower(arg[1])
		if (maybe_lvl=="debug" or maybe_lvl=="info" or maybe_lvl=="notice" or 
maybe_lvl=="warning" or maybe_lvl=="err" or maybe_lvl=="crit" or 
maybe_lvl=="alert") then
			lvl=maybe_lvl
		end
	end
	if (#arg==1) then msg=arg[1] end
	if (#arg==2) then msg=arg[2] end
	if not (isdebug and lvl=="debug") then 
freeswitch.consoleLog(lvl,scriptname .. " (" .. line .."): " .. msg .. 
"\n") end
end--]]
freeswitch.consoleLog("notice", scriptname..event:serialize("").."\n")

After restarint FS it is working and event variables are printed.

But after a while (several hours of working) FS segfaults to this:

kernel: [13834853.520624] freeswitch[31414]: segfault at 7f5b05178ca0 ip 
00007f5b08100f32 sp 00007f5ae7d32d38 error 4 in 
libc-2.11.3.so[7f5b08085000+159000]

If I comment <hook event="RECORD_STOP" script="script.lua"/>, then FS 
runs ok without segfaults.

My questions:

1) What is the better way to parse events: using start-up script which 
will hook to events and waits for them, or hook script to event in 
lua.conf and run the script only when event arise?

2) Why it segfaults on this script? Same code (function and getting 
variables) in other scripts, started from dialplan, does not segfault FS.

3) Was this a known bug and latest stable version will resolve the 
problem? I would not want to upgrade if the problem will remain.

Thank you.

-- 
Mimiko desu.


From decipher.hk at gmail.com  Sun Feb  7 21:18:27 2016
From: decipher.hk at gmail.com (=?utf-8?B?Um9kcmlnbyBSYW3DrXJleiBOb3JhbWJ1ZW5h?=)
Date: Sun, 07 Feb 2016 18:18:27 +0000
Subject: [Freeswitch-users] Answered/abandoned calls mod_callcenter
In-Reply-To: <CABMX_w-2=si4EVhFvs+7OtickCj4eMaiu+zpPwgc4RMc4Snmyg@mail.gmail.com>
References: <CABMX_w-2=si4EVhFvs+7OtickCj4eMaiu+zpPwgc4RMc4Snmyg@mail.gmail.com>
	<701581496dd4988b32ee3289cc2166df@mail2.boxtub.com>
Message-ID: <27e84038904cd4a54ac63be19844adb2@mail2.boxtub.com>

February 5 2016 3:45 PM, "?talo Rossi" <italo at freeswitch.org> wrote:
> Hi Rodrigo,

Hi ?talo,

> We don't keep these counters in memory.

Ok. 

> If you want to figure out right now how many calls were abandoned or answered you need to parse the
> cdrs from these calls or listen to the ESL events and keep a counter.
>
I see for agents we have no_answer_count and calls_answered

> But, adding realtime counters to mod_callcenter shouldn't be difficult, a PR would be awesome ;)

I'll try. 


What is the maximum line length columns?. In the [1] coding guidelines dont find 

1: https://freeswitch.org/confluence/display/FREESWITCH/Coding+Guidelines
--
Rodrigo Ram?rez Norambuena
http://www.rodrigoramirez.com


From shane.mitchell at fonedynamics.com.au  Sun Feb  7 05:37:27 2016
From: shane.mitchell at fonedynamics.com.au (Shane Mitchell)
Date: Sun, 7 Feb 2016 02:37:27 +0000
Subject: [Freeswitch-users] mod_managed: Failed to create shadow copy
Message-ID: <SG2PR04MB0791E7849B60F4D00B8FF11CCAD40@SG2PR04MB0791.apcprd04.prod.outlook.com>

Hi everyone,

I'm after a bit of mod_managed help from those with experience.

To explore how we can use mod_managed, I'm trying to simply get it up-and-running. To test, I'm trying to run Demo.csx (from source), however an ExecutionEngineException (see below) is always thrown whenever trying to load the csx file.
 
I'm using latest stable Debian Jessie and FreeSWITCH (from packages).  While I've been programming in C# since 1.0, I've never used mono, so I simply installed mono-complete from packages.  I have also tested mono (and FreeSWITCH) successfully.  I have researched related errors online with no luck or inspiration for a solution.

Does anyone know why I would get this problem? Do I need to create any symlinks, install any packages, create directories (other than conf/mod/managed), etc to get mod_managed to work on a new install?

Thanks all, much appreciated.



[ALERT] switch_cpp.cpp:1356 Exception loading /usr/lib/freeswitch/mod/managed/Demo.csx: System.ExecutionEngineException: Failed to create shadow copy (ensure directory exists).

Server stack trace:
  at (wrapper managed-to-native) System.AppDomain:LoadAssembly (System.AppDomain,string,System.Security.Policy.Evidence,bool)
  at System.AppDomain.Load (System.String assemblyString, System.Security.Policy.Evidence assemblySecurity, Boolean refonly) [0x00000] in <filename unknown>:0
  at System.AppDomain.Load (System.String assemblyString, System.Security.Policy.Evidence assemblySecurity) [0x00000] in <filename unknown>:0
  at (wrapper remoting-invoke-with-check) System.AppDomain:Load (string,System.Security.Policy.Evidence)
  at System.Reflection.Assembly.Load (System.String assemblyString, System.Security.Policy.Evidence assemblySecurity) [0x00000] in <filename unknown>:0
  at System.Activator.CreateInstance (System.String assemblyName, System.String typeName, Boolean ignoreCase, BindingFlags bindingAttr, System.Reflection.Binder binder, System.Object[] args, System.Globalization.CultureInfo culture, System.Object[] activationAttributes, System.Security.Policy.Evidence securityInfo) [0x00000] in <filename unknown>:0
  at System.Activator.CreateInstance (System.String assemblyName, System.String typeName, System.Object[] activationAttributes) [0x00000] in <filename unknown>:0
  at System.AppDomain.CreateInstance (System.String assemblyName, System.String typeName, System.Object[] activationAttributes) [0x00000] in <filename unknown>:0
  at (wrapper remoting-invoke-with-check) System.AppDomain:CreateInstance (string,string,object[])
  at System.AppDomain.CreateInstanceAndUnwrap (System.String assemblyName, System.String typeName, System.Object[] activationAttributes) [0x00000] in <filename unknown>:0
  at (wrapper remoting-invoke-with-check) System.AppDomain:CreateInstanceAndUnwrap (string,string,object[])
  at (wrapper xdomain-dispatch) System.AppDomain:CreateInstanceAndUnwrap (object,byte[]&,byte[]&,string,string)

Exception rethrown at [0]:

  at (wrapper xdomain-invoke) System.AppDomain:CreateInstanceAndUnwrap (string,string,object[])
  at (wrapper remoting-invoke-with-check) System.AppDomain:CreateInstanceAndUnwrap (string,string,object[])
  at FreeSWITCH.Loader.loadFile (System.String fileName) [0x00000] in <filename unknown>:0


From pskoul at gmail.com  Sun Feb  7 23:00:26 2016
From: pskoul at gmail.com (Panagiotis Skoulikaritis)
Date: Sun, 7 Feb 2016 22:00:26 +0200
Subject: [Freeswitch-users] extra header account code is not written to
 cdr if cancel is received a few ms after invite
In-Reply-To: <3D075DBF-8FEF-40CE-9922-A7CFC462512E@gmail.com>
References: <56AE42E1.3080703@gmail.com>
	<3CBD15A5-819B-487B-9DE0-C8120DDCAC94@gmail.com>
	<3D075DBF-8FEF-40CE-9922-A7CFC462512E@gmail.com>
Message-ID: <56B7A25A.8090801@gmail.com>

I have tried both inline and export but I still have cdrs where the
accountcode is not written.

Any help would be greatly appreciated.

Regards

Panagiotis


On 1/31/2016 11:58 PM, Oz Mortimer wrote:
> Try export rather than set
> 
>> On 31 Jan 2016, at 18:45, servtelar at gmail.com wrote:
>>
>> Shouldn't that be done as inline?
>>
>> Sent from my iPhone
>>
>>> On Jan 31, 2016, at 12:22 PM, Panagiotis Skoulikaritis <pskoul at gmail.com> wrote:
>>>
>>> Dear all
>>>
>>> I have an implementation FreeSWITCH as a sort of SBC, it is used to send
>>> the calls to the terminating carriers and do topology hiding, nothing
>>> fancy. Also I gather cdrs from the FreeSWITCH.
>>>
>>> In order to distinguish each customer on the FS cdrs I send an extra
>>> header containing the accountcode.
>>>
>>> I have noticed that if the call is canceled immediately on the same sec,
>>> the account code is not written on the cdr.
>>> To be more precise the cancel is send a few milliseconds after it has
>>> received the invite, and before the FreeSWITCH has sent the call to the
>>> terminating carrier (I'm using Homer Sipcapture to capture all the
>>> traces and I don't see an attempt being made at the terminating carrier)
>>> also I don't see a b-leg cdr.
>>>
>>> FreeSWITCH is writing both a-leg and b-leg cdrs in csv format.
>>>
>>> The dialplan that I use is simple
>>>
>>> <extension name="carrier-1">
>>> <condition field="network_addr"
>>> expression="^(^xx\.xx\.xx\.xx|^yy\.yy\.yy\.yy)$"/>
>>> <condition field="destination_number" expression="^2#(\d+)$">
>>> <action application="set" data="call_direction=outgoing"/>
>>> <action application="set" data="hangup_after_bridge=true"/>
>>> <action application="set" data="continue_on_fail=true"/>
>>> <action application="set" data="sip_cid_in_1xx=false"/>
>>> <action application="set" data="inherit_codec=true"/>
>>> <action application="set" data="accountcode=${sip_h_X-ACCOUNTCODE}"/>
>>> <action application="unset" data="sip_h_X-ACCOUNTCODE"/>
>>> <action application="set" data="execute_on_answer=sched_hangup +21600
>>> alloted_timeout"/>
>>> <action application="bridge" data="sofia/external/4567$1 at zz.zz.zz.zz:5070"/>
>>> </condition>
>>> </extension>
>>>
>>>
>>> any idea how I can make sure that the account code will always be written ?
>>>
>>>
>>> Best Regards
>>>
>>> Panagiotis
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services: 
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: 
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 covici at ccs.covici.com  Mon Feb  8 00:05:20 2016
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sun, 07 Feb 2016 16:05:20 -0500
Subject: [Freeswitch-users] mod_managed: Failed to create shadow copy
In-Reply-To: <SG2PR04MB0791E7849B60F4D00B8FF11CCAD40@SG2PR04MB0791.apcprd04.prod.outlook.com>
References: <SG2PR04MB0791E7849B60F4D00B8FF11CCAD40@SG2PR04MB0791.apcprd04.prod.outlook.com>
Message-ID: <16568.1454879120@ccs.covici.com>

Do you have a managed directory under the mod directory (not sure where
the package install puts that)?  All my .dlls are in that directory.

Shane Mitchell <shane.mitchell at fonedynamics.com.au> wrote:

> Hi everyone,
> 
> I'm after a bit of mod_managed help from those with experience.
> 
> To explore how we can use mod_managed, I'm trying to simply get it up-and-running. To test, I'm trying to run Demo.csx (from source), however an ExecutionEngineException (see below) is always thrown whenever trying to load the csx file.
>  
> I'm using latest stable Debian Jessie and FreeSWITCH (from packages).  While I've been programming in C# since 1.0, I've never used mono, so I simply installed mono-complete from packages.  I have also tested mono (and FreeSWITCH) successfully.  I have researched related errors online with no luck or inspiration for a solution.
> 
> Does anyone know why I would get this problem? Do I need to create any symlinks, install any packages, create directories (other than conf/mod/managed), etc to get mod_managed to work on a new install?
> 
> Thanks all, much appreciated.
> 
> 
> 
> [ALERT] switch_cpp.cpp:1356 Exception loading /usr/lib/freeswitch/mod/managed/Demo.csx: System.ExecutionEngineException: Failed to create shadow copy (ensure directory exists).
> 
> Server stack trace:
>   at (wrapper managed-to-native) System.AppDomain:LoadAssembly (System.AppDomain,string,System.Security.Policy.Evidence,bool)
>   at System.AppDomain.Load (System.String assemblyString, System.Security.Policy.Evidence assemblySecurity, Boolean refonly) [0x00000] in <filename unknown>:0
>   at System.AppDomain.Load (System.String assemblyString, System.Security.Policy.Evidence assemblySecurity) [0x00000] in <filename unknown>:0
>   at (wrapper remoting-invoke-with-check) System.AppDomain:Load (string,System.Security.Policy.Evidence)
>   at System.Reflection.Assembly.Load (System.String assemblyString, System.Security.Policy.Evidence assemblySecurity) [0x00000] in <filename unknown>:0
>   at System.Activator.CreateInstance (System.String assemblyName, System.String typeName, Boolean ignoreCase, BindingFlags bindingAttr, System.Reflection.Binder binder, System.Object[] args, System.Globalization.CultureInfo culture, System.Object[] activationAttributes, System.Security.Policy.Evidence securityInfo) [0x00000] in <filename unknown>:0
>   at System.Activator.CreateInstance (System.String assemblyName, System.String typeName, System.Object[] activationAttributes) [0x00000] in <filename unknown>:0
>   at System.AppDomain.CreateInstance (System.String assemblyName, System.String typeName, System.Object[] activationAttributes) [0x00000] in <filename unknown>:0
>   at (wrapper remoting-invoke-with-check) System.AppDomain:CreateInstance (string,string,object[])
>   at System.AppDomain.CreateInstanceAndUnwrap (System.String assemblyName, System.String typeName, System.Object[] activationAttributes) [0x00000] in <filename unknown>:0
>   at (wrapper remoting-invoke-with-check) System.AppDomain:CreateInstanceAndUnwrap (string,string,object[])
>   at (wrapper xdomain-dispatch) System.AppDomain:CreateInstanceAndUnwrap (object,byte[]&,byte[]&,string,string)
> 
> Exception rethrown at [0]:
> 
>   at (wrapper xdomain-invoke) System.AppDomain:CreateInstanceAndUnwrap (string,string,object[])
>   at (wrapper remoting-invoke-with-check) System.AppDomain:CreateInstanceAndUnwrap (string,string,object[])
>   at FreeSWITCH.Loader.loadFile (System.String fileName) [0x00000] in <filename unknown>:0
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 

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

         John Covici
         covici at ccs.covici.com


From shane.mitchell at fonedynamics.com.au  Mon Feb  8 02:00:35 2016
From: shane.mitchell at fonedynamics.com.au (Shane Mitchell)
Date: Sun, 7 Feb 2016 23:00:35 +0000
Subject: [Freeswitch-users] mod_managed: Failed to create shadow copy
In-Reply-To: <16568.1454879120@ccs.covici.com>
References: <SG2PR04MB0791E7849B60F4D00B8FF11CCAD40@SG2PR04MB0791.apcprd04.prod.outlook.com>
	<16568.1454879120@ccs.covici.com>
Message-ID: <SG2PR04MB079128D0A6D8A5B52D9E8185CAD40@SG2PR04MB0791.apcprd04.prod.outlook.com>

>> Do you have a managed directory under the mod directory (not sure where the package install puts that)?  All my .dlls are in that directory.

I have created the managed directory as per the initial thread.

After trial-and-error, I found the solution (simple really) to get mod_managed to work on a fresh install.  It was simply a permissions problem.  FS was trying to write to conf/mod/managed/ without sufficient permissions.

So for those looking to try mod_managed, this is what I did to get it working:

1. Install mono.
2. Create conf/mod/managed with sufficient write permissions for FS.
3. Enable/add reference to mod_managed in modules.xml.
4. Add your csx/dll/etc files to conf/mod/managed.



From stephen.thwaites at callstera.com  Mon Feb  8 14:51:52 2016
From: stephen.thwaites at callstera.com (Stephen Thwaites)
Date: Mon, 8 Feb 2016 12:51:52 +0100
Subject: [Freeswitch-users] BLF Subscriptions sometimes don't send an
	initial Notify
Message-ID: <CAB65Eivnm0iLV9o7ZLBK3FkTgTtRvguLEdY+a3ECxGKKpy5_rA@mail.gmail.com>

Hello,
I have setup presence and in most cases it is working as expected.
i.e. Subscription is sent to FS, FS returns Accepted then immediately
FS sends the notify to the phone, thereafter all the Notify events for
ringing, pickup and hangup. Great.

However in some cases a subscribe to an extension does subscribe,FS
sends the accepted response but a Notify is not sent out at that
point. However if I call the extension the Notify works perfectly.

Any ideas of what could cause the initial Notify not to be sent after
the Acceptance 202?

Any help would be appreciated.

Regards,
Steve.

p.s I sent this to the list a few days ago but it didn't seem to come
through on the list.

Some detail info below:
FS is configured as Multi-Tennant

<param name="presence-proto-lookup" value="false"/>
<param name="send-presence-on-register" value="true"/>
<param name="manage-presence" value="true"/>
<param name="presence-probe-on-register" value="true"/>
<param name="manage-shared-appearance" value="true"/>
<param name="dbname" value="freeswitch"/>
<param name="presence-hosts" value="_DISABLE_"/>   # Multi-Tennant
<param name="presence-privacy" value="false"/>

SIP Trace for Subscription

FS Receives this from the phone:

SUBSCRIBE sip:203 at xxx.mydomain.com
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>:5060
SIP/2.0
Via: SIP/2.0/UDP 192.168.1.24:25060;branch=z9hG4bK991003231;rport
From: <sip:201 at xxx.mydomain.com
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>;tag=946578510
To: <sip:203 at xxx.mydomain.com
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>;tag=nvqzSrr57RZg
Call-ID: 968288340-25060-7 at BJC.BGI.B.CE
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
CSeq: 20050 SUBSCRIBE
Contact: <sip:201 at 192.168.1.24
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>:25060>
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GXP2140 1.0.5.29
Expires: 480
Supported: replaces, path, timer, eventlist
Event: dialog
Accept: application/dialog-info+xml,multipart/related,application/rlmi+xml
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE, MESSAGE
Content-Length: 0

FS Sends this back to the phone:

SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 192.168.1.24:25060
;branch=z9hG4bK991003231;rport=59364;received=x.x.x.x
From: <sip:201 at xxx.mydomain.com
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>;tag=946578510
To: <sip:203 at xxx.mydomain.com
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>;tag=nvqzSrr57RZg
Call-ID: 968288340-25060-7 at BJC.BGI.B.CE
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
CSeq: 20050 SUBSCRIBE
Contact: <sip:203 at y.y.y.y
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>:5060>
Expires: 480
User-Agent: Callstera VOIP PBX v1.20
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer
Subscription-State: active;expires=480
Content-Length: 0
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From elvisnn at email.com  Mon Feb  8 16:26:00 2016
From: elvisnn at email.com (Elvis)
Date: Mon, 8 Feb 2016 06:26:00 -0700 (MST)
Subject: [Freeswitch-users] FreeSWITCH abandons calls
Message-ID: <1454937960255-7596200.post@n2.nabble.com>

I can see form my logs that calls are abandoned with reason [CS_NEW]
[WRONG_CALL_STATE]. Please see logs below:

[WARNING] switch_core_state_machine.c:572
2b48d9b0-ce3d-11e5-a1b4-8f0e26eefc3e
sofia/internal/61415158474 at 212.61.145.185 Abandoned
[NOTICE] switch_core_state_machine.c:575 Hangup
sofia/internal/61415158474 at 212.61.145.185 [CS_NEW] [WRONG_CALL_STATE]
[NOTICE] switch_core_session.c:1642 Session 8
(sofia/internal/61415158474 at 212.61.145.185) Ended
[NOTICE] switch_core_session.c:1646 Close Channel
sofia/internal/61415158474 at 212.61.145.185 [CS_DESTROY]


Can someone please help out?

Regards
Elvis



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-abandons-calls-tp7596200.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From brian at freeswitch.org  Mon Feb  8 18:30:08 2016
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Feb 2016 09:30:08 -0600
Subject: [Freeswitch-users] FreeSWITCH abandons calls
In-Reply-To: <1454937960255-7596200.post@n2.nabble.com>
References: <1454937960255-7596200.post@n2.nabble.com>
Message-ID: <CAEJMVkD694yj0BdUHzx-3Z_N9WnM9yHzGbcD7HHiwNC3aNFJHQ@mail.gmail.com>

This usually means that a device has sent an invite, we replied with a
challenge, but the device probably didn't receive our challenge. 'sofia
global siptrace on' and watch it.



On Mon, Feb 8, 2016 at 7:26 AM, Elvis <elvisnn at email.com> wrote:

> I can see form my logs that calls are abandoned with reason [CS_NEW]
> [WRONG_CALL_STATE]. Please see logs below:
>
> [WARNING] switch_core_state_machine.c:572
> 2b48d9b0-ce3d-11e5-a1b4-8f0e26eefc3e
> sofia/internal/61415158474 at 212.61.145.185 Abandoned
> [NOTICE] switch_core_state_machine.c:575 Hangup
> sofia/internal/61415158474 at 212.61.145.185 [CS_NEW] [WRONG_CALL_STATE]
> [NOTICE] switch_core_session.c:1642 Session 8
> (sofia/internal/61415158474 at 212.61.145.185) Ended
> [NOTICE] switch_core_session.c:1646 Close Channel
> sofia/internal/61415158474 at 212.61.145.185 [CS_DESTROY]
>
>
> Can someone please help out?
>
> Regards
> Elvis
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-abandons-calls-tp7596200.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
>



-- 

*Brian West*
brian at freeswitch.org


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

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 krice at freeswitch.org  Mon Feb  8 19:04:42 2016
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 8 Feb 2016 10:04:42 -0600
Subject: [Freeswitch-users] Verto vs. SIP.js
In-Reply-To: <1047097771.103803.1454517591330.JavaMail.zimbra@voismart.it>
References: <56AF67C9.2010700@level5.de> <95599933.20160201152539@seznam.cz>
	<003a01d15cfe$d26c3350$774499f0$@botecomm.com>
	<1105687706.20160201163013@seznam.cz>
	<505001d15d0b$623021c0$26906540$@freeswitch.org>
	<CAO6UKa2Tm5j2hF5fCnmNpgxT=7G9mpx5jagw_7PyYWAKTXQAGw@mail.gmail.com>
	<628620366.20160203092011@seznam.cz>
	<590d01d15e6a$22651810$672f4830$@freeswitch.org>
	<1047097771.103803.1454517591330.JavaMail.zimbra@voismart.it>
Message-ID: <CAPcNkdfcHqiJBBSA_iYck091q=9X4ren4-k+BrGw2e4st86cvQ@mail.gmail.com>

This ticket was already resolved

On Wed, Feb 3, 2016 at 10:39 AM, Davide Colombo <dcolombo at voismart.it>
wrote:

> I reported this bug to jira:
>
> https://freeswitch.org/jira/browse/FS-8805
>
>
>
> ----- Messaggio originale -----
> Da: "Ken Rice" <krice at freeswitch.org>
> A: "freeswitch-users" <freeswitch-users at lists.freeswitch.org>
> Inviato: Mercoled?, 3 febbraio 2016 11:03:29
> Oggetto: Re: [Freeswitch-users] Verto vs. SIP.js
>
> Re: [Freeswitch-users] Verto vs. SIP.js
>
>
> Oh So where is the Jira on this it doesn?t work in firefox with the
> debugging information?
>
>
>
> If you know of an issue like this you should report it to jira (
> https://freeswitch.org/jira ) so a dev can try to replicate and fix it?
>
>
>
> You cant expect bugs to get fixed if you aren?t reporting them properly?
>
>
>
> Go troll somewhere else?.
>
>
>
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Denis Jakovlev
> Sent: Wednesday, February 3, 2016 2:20 AM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Verto vs. SIP.js
>
>
>
>
> Dobr? den,
>
> The fact that Verto normally does not work on me Firefox does joy for me.
> For example, I have all the stops on "Refresh Media Devices" (the picture
> in the attachment)
> My clients use different browsers and I can not force them to use
> something specific.
> With sip.js I have no such problems. It works perfectly, wherever there is
> support webrtc without problems.
>
>
> --
> S pozdravem,
> Ing.Denis Jakovlev
> mob.tel. 775-415-382
>
> pond?l? 1. ?nora 2016, 17:22:24, napsal jste:
>
> With Verto you can also be more creative, see for example:
> https://freeswitch.org/verto-not-just-for-call-signaling/
> It also provides an interesting feature: "attach". e.g. if a tab is closed
> by mistake, FS can resume automatically the session upon reconnection (with
> the same ID).
> Also being a simple JSON-based protocol you can have people working on the
> client side even with limited knowledge of SIP.
>
> On 1 February 2016 at 17:12, Ken Rice < krice at freeswitch.org > wrote:
> Really? That?s why the whole protocol is OpenSource? not much vendor lock
> in there? it can be used anywhere?
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Denis
> Jakovlev
> Sent: Monday, February 1, 2016 9:30 AM
> To: FreeSWITCH Users Help < freeswitch-users at lists.freeswitch.org >
> Subject: Re: [Freeswitch-users] Verto vs. SIP.js
>
> Dobr? den,
>
> i think this Verto its too much VendorLock :)
>
> --
> S pozdravem,
> Ing.Denis Jakovlev
> mob.tel . 775-415-382
>
> pond?l? 1. ?nora 2016, 15:42:48, napsal jste:
>
> Denis, what difficulties did you experience with Verto?
>
> Thanks.
>
> Bote
>
>
> From: Denis Jakovlev
> Sent: Monday, 01 February, 2016 09:26
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Verto vs. SIP.js
>
> Dobr? den,
>
> I use sip.js. It turned out to be a lot easier that Verto.
>
> --
> S pozdravem,
> Ing.Denis Jakovlev
> mob.tel . 775-415-382
>
> pond?l? 1. ?nora 2016, 15:12:25, napsal jste:
>
> > Hi,
>
> > I want to setup a Click-2-Call-Button for our website. Is there any
> > significant difference between Verto-Mod and libraries such as SIP.js?
>
> > I do not really understand the advantage of Verto (if there is any).
>
> > Thanks in advance,
>
> > Thorsten
>
> > _________________________________________________________________________
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 DEdwards at vertical.com  Mon Feb  8 19:36:50 2016
From: DEdwards at vertical.com (Dan Edwards)
Date: Mon, 8 Feb 2016 16:36:50 +0000
Subject: [Freeswitch-users] WebSocket behind NGINX
In-Reply-To: <56B667DA.4010505@gmail.com>
References: <56AD0CE7.6000607@gmail.com> <56B667DA.4010505@gmail.com>
Message-ID: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C997CE62@PHXEX2.vertical.com>

Anton,

I'm glad my input was useful. As for WSS vs WS, the fact you're using security bubbles up into the SIP messages themselves. I initially tried:


Browser >> WSS >> Nginx >> WS >> FS


FS does not like this because the protocol changes. You go from SIP/2.0/WSS to SIP/2.0/WS and FS won't allow that. Also, in some instances, you will get SIP URL changes. For example: sip:1234 at domain.com vs. sips:1234 at domain.com.

The reason to go with WS to FS was to skip an encrypt/decrypt cycle on network traffic that never left the machine. I finally decided that trying to patch the SIP traffic was bound to fail at some point and we're only saving the encrypt/decrypt on the SIP traffic itself, so I went back to 


Browser >> WSS >> Nginx >>> WSS >> FS





-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anton
Sent: Saturday, February 06, 2016 4:39 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] WebSocket behind NGINX

Hi,

Sorry for not answering for a long time.

Dan, thank you, your recommendation really helped me.

So in order to proxy websocket request you need:
1. Proxy websocket requests in this way WSS -> (NGINX) -> FS WSS or WS 
-> (NGINX) -> FS WS
2. Modify local-network-acl
3. Modify apply-candidate-acl if you would like to drop more rtp candidates

PS: I highly recommend to watch this video about NAT issues and ACL
configuration:
https://www.youtube.com/watch?v=_WSx-T6TriI

BR,
Anton Voylenko

On 01/30/2016 09:20 PM, Anton wrote:
> Hello All,
>
> I have to proxy all websocket requests though a nginx server. Right 
> now I am using next configuration:
>
> map $http_upgrade $connection_upgrade {
>     default upgrade;
>     ''      close;
> }
>
> server {
>     listen 443;
>     server_name  wss.somedomain.com.ua;
>
>     ssl on;
>     ssl_certificate      /etc/nginx/cert.pem;
>     ssl_certificate_key  /etc/nginx/private.key;
>
>     location / {
>         proxy_pass http://127.0.0.1:5066;
>         proxy_http_version 1.1;
>         proxy_set_header Upgrade $http_upgrade;
>         proxy_set_header Connection $connection_upgrade;
>         proxy_read_timeout 86400s;
>    }
>
>     access_log /var/log/nginx/wss_access;
>     error_log /var/log/nginx/wss_error debug; }
>
> I dumped traffic from nginx and found out that "switching protocol" 
> phrase was successful but INVITE message from my browser in pending 
> state.
> Maybe FreeSWITCH wants real IP not loopback? Who have faced with 
> similar problem?
>
> BR,
> Anton


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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at 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  Mon Feb  8 22:18:42 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 8 Feb 2016 13:18:42 -0600
Subject: [Freeswitch-users] Verto vs. SIP.js
In-Reply-To: <628620366.20160203092011@seznam.cz>
References: <56AF67C9.2010700@level5.de> <95599933.20160201152539@seznam.cz>
	<003a01d15cfe$d26c3350$774499f0$@botecomm.com>
	<1105687706.20160201163013@seznam.cz>
	<505001d15d0b$623021c0$26906540$@freeswitch.org>
	<CAO6UKa2Tm5j2hF5fCnmNpgxT=7G9mpx5jagw_7PyYWAKTXQAGw@mail.gmail.com>
	<628620366.20160203092011@seznam.cz>
Message-ID: <CAKbxfG_Do-HpZ+0Ay4p8HB4SEsXmqzHRRmyU+DT3fQeQSgVJwA@mail.gmail.com>

On Wed, Feb 3, 2016 at 2:20 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:

> Dobr? den,
>
> The fact that Verto normally does not work on me Firefox does joy for me.
> For example, I have all the stops on "Refresh Media Devices" (the picture
> in the attachment)
> My clients use different browsers and I can not force them to use
> something specific.
> With sip.js I have no such problems. It works perfectly, wherever there is
> support webrtc without problems.
>
>
>
You are comparing the Verto Communicator reference app with a sip.js
library.  Apples and Oranges.   You found a bug in that app not the verto
lib.  The app developers have fixed the bug.
The best advice is keep quiet here just like you do with reporting bugs and
do not tout propaganda about one thing vs another.

The answer to OP is that either one will work.....




>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 pond?l? 1. ?nora 2016, 17:22:24, napsal jste:
> *
>
> With Verto you can also be more creative, see for example:
> https://freeswitch.org/verto-not-just-for-call-signaling/
> It also provides an interesting feature: "attach". e.g. if a tab is closed
> by mistake, FS can resume automatically the session upon reconnection (with
> the same ID).
> Also being a simple JSON-based protocol you can have people working on the
> client side even with limited knowledge of SIP.
>
> On 1 February 2016 at 17:12, Ken Rice <krice at freeswitch.org> wrote:
> Really? That?s why the whole protocol is OpenSource? not much vendor lock
> in there? it can be used anywhere?
>
>
>
> *From: *freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Denis
> Jakovlev
> *Sent:* Monday, February 1, 2016 9:30 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Verto vs. SIP.js
>
> Dobr? den,
>
> i think this Verto its too much VendorLock :)
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>
>
> *. 775-415-382 pond?l? 1. ?nora 2016, 15:42:48, napsal jste: *
>
> Denis, what difficulties did you experience with Verto?
>
> Thanks.
>
> Bote
>
>
> *From:* Denis Jakovlev
> *Sent:* Monday, 01 February, 2016 09:26
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Verto vs. SIP.js
>
> Dobr? den,
>
> I use sip.js. It turned out to be a lot easier that Verto.
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> *. 775-415-382 pond?l? 1. ?nora 2016, 15:12:25, napsal jste: > Hi, > I
> want to setup a Click-2-Call-Button for our website. Is there any >
> significant difference between Verto-Mod and libraries such as SIP.js? > I
> do not really understand the advantage of Verto (if there is any). > Thanks
> in advance, > Thorsten >
> _________________________________________________________________________*
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> <freeswitch-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 luis.daniel.lucio at gmail.com  Mon Feb  8 17:16:00 2016
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Mon, 8 Feb 2016 09:16:00 -0500
Subject: [Freeswitch-users] FreeSWITCH abandons calls
In-Reply-To: <1454937960255-7596200.post@n2.nabble.com>
References: <1454937960255-7596200.post@n2.nabble.com>
Message-ID: <CAFLo2Qx8g0caVzNT7-21qwbahKb7=9e04SoCJsRCEAmXq5Au1Q@mail.gmail.com>

Turn on debug, you will get more detailed information. Maybe answer is there
Le 8 f?vr. 2016 8:46 AM, "Elvis" <elvisnn at email.com> a ?crit :

> I can see form my logs that calls are abandoned with reason [CS_NEW]
> [WRONG_CALL_STATE]. Please see logs below:
>
> [WARNING] switch_core_state_machine.c:572
> 2b48d9b0-ce3d-11e5-a1b4-8f0e26eefc3e
> sofia/internal/61415158474 at 212.61.145.185 Abandoned
> [NOTICE] switch_core_state_machine.c:575 Hangup
> sofia/internal/61415158474 at 212.61.145.185 [CS_NEW] [WRONG_CALL_STATE]
> [NOTICE] switch_core_session.c:1642 Session 8
> (sofia/internal/61415158474 at 212.61.145.185) Ended
> [NOTICE] switch_core_session.c:1646 Close Channel
> sofia/internal/61415158474 at 212.61.145.185 [CS_DESTROY]
>
>
> Can someone please help out?
>
> Regards
> Elvis
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-abandons-calls-tp7596200.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
>
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From mbgatherer at gmail.com  Tue Feb  9 00:21:19 2016
From: mbgatherer at gmail.com (Maciej Bylica)
Date: Mon, 8 Feb 2016 22:21:19 +0100
Subject: [Freeswitch-users] Class 5 and softphone app supporting ZRTP
Message-ID: <CAEj_Gpj=xBtC3DduB7EQS4_zrv=BVPpvHSTUue8dmBdSU8n1zg@mail.gmail.com>

Hi All,

I am looking for a class 5 platform (basic VAS) and softphone (IOS,
Android) both supporting ZRTP protocol to achieve the highest voice
security.
C.5 and UA should be delivered from the same supplier (like sipwise for
instance)

Could anybody recommend me any solution here?

Thanks in advanced
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From rtreleaven at bunnykick.ca  Tue Feb  9 00:49:58 2016
From: rtreleaven at bunnykick.ca (Russell Treleaven)
Date: Mon, 8 Feb 2016 16:49:58 -0500
Subject: [Freeswitch-users] Class 5 and softphone app supporting ZRTP
In-Reply-To: <CAEj_Gpj=xBtC3DduB7EQS4_zrv=BVPpvHSTUue8dmBdSU8n1zg@mail.gmail.com>
References: <CAEj_Gpj=xBtC3DduB7EQS4_zrv=BVPpvHSTUue8dmBdSU8n1zg@mail.gmail.com>
Message-ID: <CAHzfiZ4eru_kVJRfvqCHhrAYcHfxdyKtVuo8AsdzGoGHVeLfQg@mail.gmail.com>

This is the second mention of Class 5 in last couple of months.
If you don't mind my asking why is Class 5 a requirement?
Put another way is their a public specification that we should be
referencing?

Sincerely,

Russell Treleaven


On Mon, Feb 8, 2016 at 4:21 PM, Maciej Bylica <mbgatherer at gmail.com> wrote:

> Hi All,
>
> I am looking for a class 5 platform (basic VAS) and softphone (IOS,
> Android) both supporting ZRTP protocol to achieve the highest voice
> security.
> C.5 and UA should be delivered from the same supplier (like sipwise for
> instance)
>
> Could anybody recommend me any solution here?
>
> Thanks in advanced
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mbgatherer at gmail.com  Tue Feb  9 01:15:48 2016
From: mbgatherer at gmail.com (Maciej Bylica)
Date: Mon, 8 Feb 2016 23:15:48 +0100
Subject: [Freeswitch-users] Class 5 and softphone app supporting ZRTP
In-Reply-To: <CAHzfiZ4eru_kVJRfvqCHhrAYcHfxdyKtVuo8AsdzGoGHVeLfQg@mail.gmail.com>
References: <CAEj_Gpj=xBtC3DduB7EQS4_zrv=BVPpvHSTUue8dmBdSU8n1zg@mail.gmail.com>
	<CAHzfiZ4eru_kVJRfvqCHhrAYcHfxdyKtVuo8AsdzGoGHVeLfQg@mail.gmail.com>
Message-ID: <CAEj_GpjO3KeyjR8zK_Xs2oeYejLxmMQ2afQLtxC5iBbBhbkZLQ@mail.gmail.com>

Hi

Thanks for prompt answer.

By saying class5, i was referring to platform capable of delivering VoIP
features to the end subscribers.
Of course it could be FS+OS/Kamailio+DB+scripting+web

As for the beginning i need basic features like Voicemail, IVR, Call Hold,
CallWaiting, but wouldn't like to limit myself in the future.
The app needs to be auto-provisioned once downloaded and installed. Push
notification, in-app purchases and ZRTP is a must here.

Thanks
Maciej.


2016-02-08 22:49 GMT+01:00 Russell Treleaven <rtreleaven at bunnykick.ca>:

> This is the second mention of Class 5 in last couple of months.
> If you don't mind my asking why is Class 5 a requirement?
> Put another way is their a public specification that we should be
> referencing?
>
> Sincerely,
>
> Russell Treleaven
>
>
> On Mon, Feb 8, 2016 at 4:21 PM, Maciej Bylica <mbgatherer at gmail.com>
> wrote:
>
>> Hi All,
>>
>> I am looking for a class 5 platform (basic VAS) and softphone (IOS,
>> Android) both supporting ZRTP protocol to achieve the highest voice
>> security.
>> C.5 and UA should be delivered from the same supplier (like sipwise for
>> instance)
>>
>> Could anybody recommend me any solution here?
>>
>> Thanks in advanced
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 netcentrica at gmail.com  Tue Feb  9 01:40:42 2016
From: netcentrica at gmail.com (Mateusz Bartczak)
Date: Mon, 8 Feb 2016 23:40:42 +0100
Subject: [Freeswitch-users] Enterprise/sequential originate different stop
	codes per gateway
Message-ID: <CAJWEF2YZ=FZ5+YE_Lduhp_2JxC57HtmesaqhpFHhXYKbi95syQ@mail.gmail.com>

Hi all

I'm looking for a proper way to achieve functionality of different stop
codes per every called gateway

Example call flow:

call provider 1, stop on code 404
if provider 1 fails call provider 2, stop on code 404,486
if provider 2 fails call provider 3, stop on code 480
and so on

I'm aware of variable continue_on_fail but I had no luck in specifying this
per gateway, it seems to work only for whole originate string, not
individual gateways

Could you share some examples how to do this?

Best Regards
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From s.safarov at gmail.com  Tue Feb  9 07:41:01 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 9 Feb 2016 07:41:01 +0300
Subject: [Freeswitch-users] FreeSWITCH abandons calls
In-Reply-To: <CAFLo2Qx8g0caVzNT7-21qwbahKb7=9e04SoCJsRCEAmXq5Au1Q@mail.gmail.com>
References: <1454937960255-7596200.post@n2.nabble.com>
	<CAFLo2Qx8g0caVzNT7-21qwbahKb7=9e04SoCJsRCEAmXq5Au1Q@mail.gmail.com>
Message-ID: <CAHtxdDfY2QtKpUPkBROSGtZmaoPMcynZLNuyYvpfaceDRXuifA@mail.gmail.com>

Probably you server has public IP address.
In this case hackers searching open servers or account with out password
registration.
FS-7125 <https://freeswitch.org/jira/browse/FS-7125>
PR-159
<https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/159/overview>


On Mon, Feb 8, 2016 at 5:16 PM, Luis Daniel Lucio Quiroz <
luis.daniel.lucio at gmail.com> wrote:

> Turn on debug, you will get more detailed information. Maybe answer is
> there
> Le 8 f?vr. 2016 8:46 AM, "Elvis" <elvisnn at email.com> a ?crit :
>
>> I can see form my logs that calls are abandoned with reason [CS_NEW]
>> [WRONG_CALL_STATE]. Please see logs below:
>>
>> [WARNING] switch_core_state_machine.c:572
>> 2b48d9b0-ce3d-11e5-a1b4-8f0e26eefc3e
>> sofia/internal/61415158474 at 212.61.145.185 Abandoned
>> [NOTICE] switch_core_state_machine.c:575 Hangup
>> sofia/internal/61415158474 at 212.61.145.185 [CS_NEW] [WRONG_CALL_STATE]
>> [NOTICE] switch_core_session.c:1642 Session 8
>> (sofia/internal/61415158474 at 212.61.145.185) Ended
>> [NOTICE] switch_core_session.c:1646 Close Channel
>> sofia/internal/61415158474 at 212.61.145.185 [CS_DESTROY]
>>
>>
>> Can someone please help out?
>>
>> Regards
>> Elvis
>>
>>
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-abandons-calls-tp7596200.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
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 idokan at gmail.com  Tue Feb  9 08:42:22 2016
From: idokan at gmail.com (ik)
Date: Tue, 9 Feb 2016 07:42:22 +0200
Subject: [Freeswitch-users] Capturing leg B hangup from bridge
Message-ID: <CAOccsKCzsbJMkfspmBU7-DRSx43aZ-3O8dTRxo1h9d13rYMPFA@mail.gmail.com>

Hello,

I'm trying to use hangup api hook for leg b, while leg a is still connected
and transferred to do other tasks.

The hook only triggers when leg a disconnected, not when leg b.

I found this question on Google, but no answer that works.

So how can I execute API (curl in this case) when leg b is disconnected?

Thank you,

Ido
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From idokan at gmail.com  Tue Feb  9 11:09:54 2016
From: idokan at gmail.com (ik)
Date: Tue, 9 Feb 2016 10:09:54 +0200
Subject: [Freeswitch-users] Capturing leg B hangup from bridge
In-Reply-To: <CAOccsKCzsbJMkfspmBU7-DRSx43aZ-3O8dTRxo1h9d13rYMPFA@mail.gmail.com>
References: <CAOccsKCzsbJMkfspmBU7-DRSx43aZ-3O8dTRxo1h9d13rYMPFA@mail.gmail.com>
Message-ID: <CAOccsKAgs1EtS5t=Ozr9w_q9VCo1JraTd9h+gqgAjeoNwtzZPA@mail.gmail.com>

Answering myself : export and nolocal to do it.
On Feb 9, 2016 7:42 AM, "ik" <idokan at gmail.com> wrote:

> Hello,
>
> I'm trying to use hangup api hook for leg b, while leg a is still
> connected and transferred to do other tasks.
>
> The hook only triggers when leg a disconnected, not when leg b.
>
> I found this question on Google, but no answer that works.
>
> So how can I execute API (curl in this case) when leg b is disconnected?
>
> Thank you,
>
> Ido
>
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From achinthau at gmail.com  Tue Feb  9 15:39:01 2016
From: achinthau at gmail.com (Achintha)
Date: Tue, 9 Feb 2016 18:09:01 +0530
Subject: [Freeswitch-users] video conference not working
Message-ID: <CA+Jwzxv5dSWNfWPg8JZOY6JOp7bHmaPZ6X5+yj7ZzMpKtR4-QQ@mail.gmail.com>

hi all,


We are checking SIP video conferencing on FreeSwitch through mod_conference
but the video is not getting enabled. Video works fine on a normal user to
user call and codecs are matching. But for the conference room we create,
the softphone displays an error message saying ?No matching video codecs
found?. We are using FreeSwitch version 1.6
kindly advice me to solve this problem

Thanking you
Achintha
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From gmaruzz at gmail.com  Tue Feb  9 16:21:33 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 9 Feb 2016 14:21:33 +0100
Subject: [Freeswitch-users] video conference not working
In-Reply-To: <CA+Jwzxv5dSWNfWPg8JZOY6JOp7bHmaPZ6X5+yj7ZzMpKtR4-QQ@mail.gmail.com>
References: <CA+Jwzxv5dSWNfWPg8JZOY6JOp7bHmaPZ6X5+yj7ZzMpKtR4-QQ@mail.gmail.com>
Message-ID: <CALXCt0rZyGrkj3qZGSih2obKgCnAZY3O2vXC=T5paZ6-EdUA+Q@mail.gmail.com>

you must pastebin the dialplan, the conference settings, the complete debug
from fs_cli

how we can know what is happening?



On Tue, Feb 9, 2016 at 1:39 PM, Achintha <achinthau at gmail.com> wrote:

> hi all,
>
>
> We are checking SIP video conferencing on FreeSwitch through
> mod_conference but the video is not getting enabled. Video works fine on a
> normal user to user call and codecs are matching. But for the conference
> room we create, the softphone displays an error message saying ?No matching
> video codecs found?. We are using FreeSwitch version 1.6
> kindly advice me to solve this problem
>
> Thanking you
> Achintha
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 elvisnn at email.com  Tue Feb  9 16:37:02 2016
From: elvisnn at email.com (Elvis N. Ngah)
Date: Tue, 9 Feb 2016 14:37:02 +0100
Subject: [Freeswitch-users] FreeSWITCH abandons calls
In-Reply-To: <CAEJMVkD694yj0BdUHzx-3Z_N9WnM9yHzGbcD7HHiwNC3aNFJHQ@mail.gmail.com>
References: <1454937960255-7596200.post@n2.nabble.com>,
	<CAEJMVkD694yj0BdUHzx-3Z_N9WnM9yHzGbcD7HHiwNC3aNFJHQ@mail.gmail.com>
Message-ID: <trinity-76f5ac22-52ff-478a-ad4b-4a7ec67a4030-1455025022331@3capp-mailcom-lxa10>

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From brian at freeswitch.org  Tue Feb  9 16:42:48 2016
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2016 07:42:48 -0600
Subject: [Freeswitch-users] FreeSWITCH abandons calls
In-Reply-To: <trinity-76f5ac22-52ff-478a-ad4b-4a7ec67a4030-1455025022331@3capp-mailcom-lxa10>
References: <1454937960255-7596200.post@n2.nabble.com>
	<CAEJMVkD694yj0BdUHzx-3Z_N9WnM9yHzGbcD7HHiwNC3aNFJHQ@mail.gmail.com>
	<trinity-76f5ac22-52ff-478a-ad4b-4a7ec67a4030-1455025022331@3capp-mailcom-lxa10>
Message-ID: <CAEJMVkAtGwxmy_ORSY0+-ZQcoA99r+jS8MbU_rEBEDfB3Jh6yw@mail.gmail.com>

You've got a nat problem, disabling this won't probably fix your issue!
What devices are involved and network topology?

/b

On Tuesday, February 9, 2016, Elvis N. Ngah <elvisnn at email.com> wrote:

> Thank you Brian!
>
> I did so and saw exactly what you told me. The INVITE is being challenged
> with a 407 - Proxy Authentication Required. I would like to disable this
> feature. Can you please let me know where to do that?
>
> Regards
> Elvis
>
> *Sent:* Monday, February 08, 2016 at 4:30 PM
> *From:* "Brian West" <brian at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','brian at freeswitch.org');>>
> *To:* "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users at lists.freeswitch.org');>>
> *Subject:* Re: [Freeswitch-users] FreeSWITCH abandons calls
> This usually means that a device has sent an invite, we replied with a
> challenge, but the device probably didn't receive our challenge. 'sofia
> global siptrace on' and watch it.
>
>
>
> On Mon, Feb 8, 2016 at 7:26 AM, Elvis <elvisnn at email.com> wrote:
>>
>> I can see form my logs that calls are abandoned with reason [CS_NEW]
>> [WRONG_CALL_STATE]. Please see logs below:
>>
>> [WARNING] switch_core_state_machine.c:572
>> 2b48d9b0-ce3d-11e5-a1b4-8f0e26eefc3e
>> sofia/internal/61415158474 at 212.61.145.185 Abandoned
>> [NOTICE] switch_core_state_machine.c:575 Hangup
>> sofia/internal/61415158474 at 212.61.145.185 [CS_NEW] [WRONG_CALL_STATE]
>> [NOTICE] switch_core_session.c:1642 Session 8
>> (sofia/internal/61415158474 at 212.61.145.185) Ended
>> [NOTICE] switch_core_session.c:1646 Close Channel
>> sofia/internal/61415158474 at 212.61.145.185 [CS_DESTROY]
>>
>>
>> Can someone please help out?
>>
>> Regards
>> Elvis
>>
>>
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-abandons-calls-tp7596200.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
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> 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
> <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


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

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 bilaln018 at gmail.com  Tue Feb  9 16:51:51 2016
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Tue, 9 Feb 2016 18:51:51 +0500
Subject: [Freeswitch-users] [WSS Unable to connect][No Debug logs]
Message-ID: <CADSVerCK0YBSAibYCttUwz70yMMntiMGxCDGOO6s+1mrQk08oA@mail.gmail.com>

Hi all,
I just want to enable the wss on freeswitch, so i have configuration in
place, wss listening on port 7443, wss.pem is added in the certs directory,
but i am unable to connect using https://tryit.jssip.net/,(i can get
registered using wss://tryit.jssip.net:10443 but not through
wss://MYSERVERIP:7443)
So my question is how can i check the debug logs that whats actually
happening on the switch, i can see the hits coming on server using tcpdump.
Any help will be highly appreciated.

Thanks
Abbasi
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From brian at freeswitch.org  Tue Feb  9 17:14:40 2016
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Feb 2016 08:14:40 -0600
Subject: [Freeswitch-users] [WSS Unable to connect][No Debug logs]
In-Reply-To: <CADSVerCK0YBSAibYCttUwz70yMMntiMGxCDGOO6s+1mrQk08oA@mail.gmail.com>
References: <CADSVerCK0YBSAibYCttUwz70yMMntiMGxCDGOO6s+1mrQk08oA@mail.gmail.com>
Message-ID: <CAEJMVkAxT59yizWi1XnR5aXMdqtZ6BW0rPp+5sG1FuF2qpht=Q@mail.gmail.com>

Did you setup your certificates? if you try to visit https://MYSERVERIP:7443
what do you get?

On Tue, Feb 9, 2016 at 7:51 AM, Bilal Abbasi <bilaln018 at gmail.com> wrote:

> Hi all,
> I just want to enable the wss on freeswitch, so i have configuration in
> place, wss listening on port 7443, wss.pem is added in the certs directory,
> but i am unable to connect using https://tryit.jssip.net/,(i can get
> registered using wss://tryit.jssip.net:10443 but not through
> wss://MYSERVERIP:7443)
> So my question is how can i check the debug logs that whats actually
> happening on the switch, i can see the hits coming on server using tcpdump.
> Any help will be highly appreciated.
>
> Thanks
> Abbasi
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


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

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 DEdwards at vertical.com  Tue Feb  9 17:18:01 2016
From: DEdwards at vertical.com (Dan Edwards)
Date: Tue, 9 Feb 2016 14:18:01 +0000
Subject: [Freeswitch-users] [WSS Unable to connect][No Debug logs]
In-Reply-To: <CADSVerCK0YBSAibYCttUwz70yMMntiMGxCDGOO6s+1mrQk08oA@mail.gmail.com>
References: <CADSVerCK0YBSAibYCttUwz70yMMntiMGxCDGOO6s+1mrQk08oA@mail.gmail.com>
Message-ID: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C997D3D1@PHXEX2.vertical.com>

If the certificates in wss.pem are not created for MYSERVERIP, this will fail.

The easiest way to check is to enter the URL in your browsers? address bar, substituting https for wss (ie. https://MYSERVERIP:7443). If your browser throws up a security issue, the SSL certificates in wss.pem do not match your host. If that?s the case, you?ll need to purchase your own SSL certificates (if you want to make this publically available) or create self-signed certs for testing/debugging.

See https://freeswitch.org/confluence/display/FREESWITCH/WebRTC for more info.


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bilal Abbasi
Sent: Tuesday, February 09, 2016 8:52 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] [WSS Unable to connect][No Debug logs]

Hi all,
I just want to enable the wss on freeswitch, so i have configuration in place, wss listening on port 7443, wss.pem is added in the certs directory, but i am unable to connect using https://tryit.jssip.net/,(i can get registered using wss://tryit.jssip.net:10443<http://tryit.jssip.net:10443> but not through wss://MYSERVERIP:7443)
So my question is how can i check the debug logs that whats actually happening on the switch, i can see the hits coming on server using tcpdump.
Any help will be highly appreciated.

Thanks
Abbasi
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From astpp at inextrix.com  Mon Feb  8 12:44:57 2016
From: astpp at inextrix.com (ASTPP Opensource VOIP Billing)
Date: Mon, 8 Feb 2016 15:14:57 +0530
Subject: [Freeswitch-users] ASTPP Team launched Mobile SIP Dialer
Message-ID: <CAP0=6-Aud2At8-5d_ctoaypmxmf61KAWS6-tfOE2fntv8ffoOA@mail.gmail.com>

Hi Everyone,

ASTPP Team glad to announce first ever Mobile SIP Dialer which is
integrated with ASTPP.

*ASTPP Dialer* is a brand new Mobile SIP Dialer launched by ASTPP Team. Its
a complete SIP Softphone which allows you to register your SIP account on
any SIP server, specially ASTPP server.


*Features:*

   - Easy to use & User friendly graphical interface
   - Call Transfer, Call Hold
   - Integration with Mobile Phonebook
   - Fast registration and call connectivity
   - Works with any SIP server
   - Displays real time Account balance (ASTPP only)
   - Displays Call history
   - Connect through 3G, 4G, GPRS and Wi-Fi
   - G729, GSM, iLBC, Speex, G711, G722, AMR Codec Support


*Download:*

ASTPP Dialer is available freely in *Google play store*:
https://play.google.com/store/apps/details?id=com.inextrix.astppdialer

Please do not forget to share your review in play store.


*Want your own branded ASTPP Dialer?*

We do also offer complete integration with your ASTPP server, and *rebrand
/ customize it with your company name/logo* in both *Android and iOS*. To
get more details on it please *Click Here
<http://www.astppbilling.org/astpp-mobile-dialer-transformation> *and send
your requirements on sales at inextrix.com

-- 
Regards,
ASTPP Team
iNextrix Technologies Pvt. Ltd.
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From yadenis at seznam.cz  Tue Feb  9 10:47:26 2016
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Tue, 9 Feb 2016 08:47:26 +0100
Subject: [Freeswitch-users] Verto vs. SIP.js
In-Reply-To: <CAKbxfG_Do-HpZ+0Ay4p8HB4SEsXmqzHRRmyU+DT3fQeQSgVJwA@mail.gmail.com>
References: <56AF67C9.2010700@level5.de> <95599933.20160201152539@seznam.cz> 
	<003a01d15cfe$d26c3350$774499f0$@botecomm.com>
	<1105687706.20160201163013@seznam.cz>
	<505001d15d0b$623021c0$26906540$@freeswitch.org>
	<CAO6UKa2Tm5j2hF5fCnmNpgxT=7G9mpx5jagw_7PyYWAKTXQAGw@mail.gmail.com>
	<628620366.20160203092011@seznam.cz>
	<CAKbxfG_Do-HpZ+0Ay4p8HB4SEsXmqzHRRmyU+DT3fQeQSgVJwA@mail.gmail.com>
Message-ID: <1481152288.20160209084726@seznam.cz>

Dobr? den,

I'm sorry, but in order to try, as it generally works I try first reference app. If this does not work where I want, I just go on. No propaganda. For a button to call a site I stopped at sip.js. It's all. If I can not say this when asked, I apologize. More I will not do.

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

pond?l? 8. ?nora 2016, 20:18:42, napsal jste:



On Wed, Feb 3, 2016 at 2:20 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:
Dobr? den,

The fact that Verto normally does not work on me Firefox does joy for me. For example, I have all the stops on "Refresh Media Devices" (the picture in the attachment)
My clients use different browsers and I can not force them to use something specific. 
With sip.js I have no such problems. It works perfectly, wherever there is support webrtc without problems.



You are comparing the Verto Communicator reference app with a sip.js library.  Apples and Oranges.   You found a bug in that app not the verto lib.  The app developers have fixed the bug.
The best advice is keep quiet here just like you do with reporting bugs and do not tout propaganda about one thing vs another.

The answer to OP is that either one will work.....


 
-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

pond?l? 1. ?nora 2016, 17:22:24, napsal jste:


With Verto you can also be more creative, see for example: https://freeswitch.org/verto-not-just-for-call-signaling/
It also provides an interesting feature: "attach". e.g. if a tab is closed by mistake, FS can resume automatically the session upon reconnection (with the same ID).
Also being a simple JSON-based protocol you can have people working on the client side even with limited knowledge of SIP.

On 1 February 2016 at 17:12, Ken Rice <krice at freeswitch.org> wrote:
Really? That?s why the whole protocol is OpenSource? not much vendor lock in there? it can be used anywhere? 



From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Denis Jakovlev
Sent: Monday, February 1, 2016 9:30 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Verto vs. SIP.js

Dobr? den,

i think this Verto its too much VendorLock :) 

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

pond?l? 1. ?nora 2016, 15:42:48, napsal jste:

Denis, what difficulties did you experience with Verto?

Thanks.

Bote


From: Denis Jakovlev
Sent: Monday, 01 February, 2016 09:26
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Verto vs. SIP.js

Dobr? den,

I use sip.js. It turned out to be a lot easier that Verto.

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

pond?l? 1. ?nora 2016, 15:12:25, napsal jste:

> Hi,

> I want to setup a Click-2-Call-Button for our website. Is there any 
> significant difference between Verto-Mod and libraries such as SIP.js?

> I do not really understand the advantage of Verto (if there is any).

> Thanks in advance,

> Thorsten

> _________________________________________________________________________


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

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

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

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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at 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
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From elvisnn at email.com  Tue Feb  9 17:47:22 2016
From: elvisnn at email.com (Elvis N. Ngah)
Date: Tue, 9 Feb 2016 15:47:22 +0100
Subject: [Freeswitch-users] FreeSWITCH abandons calls
In-Reply-To: <CAEJMVkAtGwxmy_ORSY0+-ZQcoA99r+jS8MbU_rEBEDfB3Jh6yw@mail.gmail.com>
References: <1454937960255-7596200.post@n2.nabble.com>
	<CAEJMVkD694yj0BdUHzx-3Z_N9WnM9yHzGbcD7HHiwNC3aNFJHQ@mail.gmail.com>
	<trinity-76f5ac22-52ff-478a-ad4b-4a7ec67a4030-1455025022331@3capp-mailcom-lxa10>,
	<CAEJMVkAtGwxmy_ORSY0+-ZQcoA99r+jS8MbU_rEBEDfB3Jh6yw@mail.gmail.com>
Message-ID: <trinity-ea4211ac-41d4-4182-a47b-42073b3073dd-1455029241963@3capp-mailcom-lxa10>

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From royj at yandex.ru  Tue Feb  9 18:10:29 2016
From: royj at yandex.ru (royj at yandex.ru)
Date: Tue, 09 Feb 2016 18:10:29 +0300
Subject: [Freeswitch-users] Outbound async ESL, linger, pickup
Message-ID: <848891455030629@web25h.yandex.ru>

Hi, all

Is anybody used pickup ' https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+pickup ' in outbound async ESL, linger?

When trying pickup for example after this:

sendmsg
execute-app-name: bridge
execute-app-arg: sofia/gateway/gateway_name/number,pickup/100
call-command: execute

with:

sendmsg
execute-app-name: pickup
execute-app-arg: 100
call-command: execute

I got +OK and immediately CHANNEL_EXECUTE_COMPLETE, CHANNEL_HANGUP_COMPLETE for channel that calls pickup.

It is not a script issues hangup command.
But if that pickup to call from dialplan like <action application="pickup" data="100"/> and bridge from ESL then all works as expected.

Can somebody point what might be wrong here? Or clarify the picture.


From gmaruzz at gmail.com  Tue Feb  9 20:25:53 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 9 Feb 2016 18:25:53 +0100
Subject: [Freeswitch-users] Hit that wallet now ! FreeSWITCH core lunch and
	dinner !
Message-ID: <CALXCt0r3MA37Tv_GtULEysa3kQfOC_e-xYvNAYZJKpixs0tt5A@mail.gmail.com>

Hey fellow FreeSWITCHers!

How much we all made in 2015 thanks to FreeSWITCH?

How much we made thanks to core devs (Anthony, Mike, Ken, Brian, William,
etc) answering our difficult questions?

Let's send them MONEY so they can have a LUXURIOUS lunch and dinner on us !

On top right in http://www.freeswitch.org page there is a "Donate" Paypal
button that will directly translate in gluttony for the Core FreeSWITCH
Team.

Core devs are having the annual development meeting days.

If not now, when?

Because!

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From blake at cogents.io  Tue Feb  9 21:26:45 2016
From: blake at cogents.io (Blake Priddy)
Date: Tue, 9 Feb 2016 12:26:45 -0600
Subject: [Freeswitch-users] Hit that wallet now ! FreeSWITCH core lunch
 and dinner !
In-Reply-To: <CALXCt0r3MA37Tv_GtULEysa3kQfOC_e-xYvNAYZJKpixs0tt5A@mail.gmail.com>
References: <CALXCt0r3MA37Tv_GtULEysa3kQfOC_e-xYvNAYZJKpixs0tt5A@mail.gmail.com>
Message-ID: <CA+Kn03=nx1C7WpsmAEgzXBSVWBR1q-KCFkw3KOSHA2YzSgyaTw@mail.gmail.com>

C'mon y'all!!!

https://www.gofundme.com/freeswitch
On Feb 9, 2016 12:22 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:

> Hey fellow FreeSWITCHers!
>
> How much we all made in 2015 thanks to FreeSWITCH?
>
> How much we made thanks to core devs (Anthony, Mike, Ken, Brian, William,
> etc) answering our difficult questions?
>
> Let's send them MONEY so they can have a LUXURIOUS lunch and dinner on us !
>
> On top right in http://www.freeswitch.org page there is a "Donate" Paypal
> button that will directly translate in gluttony for the Core FreeSWITCH
> Team.
>
> Core devs are having the annual development meeting days.
>
> If not now, when?
>
> Because!
>
> -giovanni
>
> --
> 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 thomas at bettervoice.com  Wed Feb 10 05:23:17 2016
From: thomas at bettervoice.com (Thomas Quintana)
Date: Tue, 9 Feb 2016 21:23:17 -0500
Subject: [Freeswitch-users] Hit that wallet now ! FreeSWITCH core lunch
 and dinner !
In-Reply-To: <CA+Kn03=nx1C7WpsmAEgzXBSVWBR1q-KCFkw3KOSHA2YzSgyaTw@mail.gmail.com>
References: <CALXCt0r3MA37Tv_GtULEysa3kQfOC_e-xYvNAYZJKpixs0tt5A@mail.gmail.com>
	<CA+Kn03=nx1C7WpsmAEgzXBSVWBR1q-KCFkw3KOSHA2YzSgyaTw@mail.gmail.com>
Message-ID: <CAJ=fo8xvXcAS9PmEYDZ=pJ3rfp-8HNMZARaE-N7J6ndNLpgWZg@mail.gmail.com>

+1

Thomas Quintana
Chief Technology Officer
Phone: 1 (512) 677-6200
Website: http://www.bettervoice.com



On Tue, Feb 9, 2016 at 1:26 PM, Blake Priddy <blake at cogents.io> wrote:

> C'mon y'all!!!
>
> https://www.gofundme.com/freeswitch
> On Feb 9, 2016 12:22 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:
>
>> Hey fellow FreeSWITCHers!
>>
>> How much we all made in 2015 thanks to FreeSWITCH?
>>
>> How much we made thanks to core devs (Anthony, Mike, Ken, Brian, William,
>> etc) answering our difficult questions?
>>
>> Let's send them MONEY so they can have a LUXURIOUS lunch and dinner on us
>> !
>>
>> On top right in http://www.freeswitch.org page there is a "Donate"
>> Paypal button that will directly translate in gluttony for the Core
>> FreeSWITCH Team.
>>
>> Core devs are having the annual development meeting days.
>>
>> If not now, when?
>>
>> Because!
>>
>> -giovanni
>>
>> --
>> 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 bilaln018 at gmail.com  Wed Feb 10 08:03:17 2016
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Wed, 10 Feb 2016 10:03:17 +0500
Subject: [Freeswitch-users] [WSS Unable to connect][No Debug logs]
In-Reply-To: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C997D3D1@PHXEX2.vertical.com>
References: <CADSVerCK0YBSAibYCttUwz70yMMntiMGxCDGOO6s+1mrQk08oA@mail.gmail.com>
	<1ECCD30B07AF6E4EB58BC5F1E9FFF981C997D3D1@PHXEX2.vertical.com>
Message-ID: <CADSVerA5Pip+hr3ntgov=L7B0qDNVKZszn+S1gZEv+39OFr2oQ@mail.gmail.com>

Hi Brain,Edwards,
Thanks a lot for the comments, yes the issue was with wss.pem file, i have
purchased the domain and certificate for my server, and it started working,

Regards
Abbasi

On Tue, Feb 9, 2016 at 7:18 PM, Dan Edwards <DEdwards at vertical.com> wrote:

> If the certificates in wss.pem are not created for MYSERVERIP, this will
> fail.
>
>
>
> The easiest way to check is to enter the URL in your browsers? address
> bar, substituting https for wss (ie. https://MYSERVERIP:7443). If your
> browser throws up a security issue, the SSL certificates in wss.pem do not
> match your host. If that?s the case, you?ll need to purchase your own SSL
> certificates (if you want to make this publically available) or create
> self-signed certs for testing/debugging.
>
>
>
> See https://freeswitch.org/confluence/display/FREESWITCH/WebRTC for more
> info.
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Bilal Abbasi
> *Sent:* Tuesday, February 09, 2016 8:52 AM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] [WSS Unable to connect][No Debug logs]
>
>
>
> Hi all,
>
> I just want to enable the wss on freeswitch, so i have configuration in
> place, wss listening on port 7443, wss.pem is added in the certs directory,
> but i am unable to connect using https://tryit.jssip.net/,(i can get
> registered using wss://tryit.jssip.net:10443 but not through
> wss://MYSERVERIP:7443)
>
> So my question is how can i check the debug logs that whats actually
> happening on the switch, i can see the hits coming on server using tcpdump.
>
> Any help will be highly appreciated.
>
>
>
> Thanks
>
> Abbasi
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jprangi at didforsale.com  Wed Feb 10 09:55:39 2016
From: jprangi at didforsale.com (Jai Rangi)
Date: Tue, 9 Feb 2016 22:55:39 -0800
Subject: [Freeswitch-users] Hit that wallet now ! FreeSWITCH core lunch
 and dinner !
In-Reply-To: <CAJ=fo8xvXcAS9PmEYDZ=pJ3rfp-8HNMZARaE-N7J6ndNLpgWZg@mail.gmail.com>
References: <CALXCt0r3MA37Tv_GtULEysa3kQfOC_e-xYvNAYZJKpixs0tt5A@mail.gmail.com>
	<CA+Kn03=nx1C7WpsmAEgzXBSVWBR1q-KCFkw3KOSHA2YzSgyaTw@mail.gmail.com>
	<CAJ=fo8xvXcAS9PmEYDZ=pJ3rfp-8HNMZARaE-N7J6ndNLpgWZg@mail.gmail.com>
Message-ID: <CAPfjz3hzBQkyB42-BxV5xREa0U4ToXe-rtNDeicYS-itoh8ULw@mail.gmail.com>

+2

*Jai Rangi*
Cebod Technologies LLC dba DIDforSale/Cebod Telecom
O 949-471-0102 <O%20949-471-0101> | C 949-419-7634 <C%20949-742-2666> | F
949-269-0449 / 949-232-1410 | jprangi at didforsale.com  www.cebod.com |
www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA 92626  |






On Tue, Feb 9, 2016 at 6:23 PM, Thomas Quintana <thomas at bettervoice.com>
wrote:

> +1
>
> Thomas Quintana
> Chief Technology Officer
> Phone: 1 (512) 677-6200
> Website: http://www.bettervoice.com
>
>
>
> On Tue, Feb 9, 2016 at 1:26 PM, Blake Priddy <blake at cogents.io> wrote:
>
>> C'mon y'all!!!
>>
>> https://www.gofundme.com/freeswitch
>> On Feb 9, 2016 12:22 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:
>>
>>> Hey fellow FreeSWITCHers!
>>>
>>> How much we all made in 2015 thanks to FreeSWITCH?
>>>
>>> How much we made thanks to core devs (Anthony, Mike, Ken, Brian,
>>> William, etc) answering our difficult questions?
>>>
>>> Let's send them MONEY so they can have a LUXURIOUS lunch and dinner on
>>> us !
>>>
>>> On top right in http://www.freeswitch.org page there is a "Donate"
>>> Paypal button that will directly translate in gluttony for the Core
>>> FreeSWITCH Team.
>>>
>>> Core devs are having the annual development meeting days.
>>>
>>> If not now, when?
>>>
>>> Because!
>>>
>>> -giovanni
>>>
>>> --
>>> 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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 blake at cogents.io  Wed Feb 10 11:07:10 2016
From: blake at cogents.io (Blake Priddy)
Date: Wed, 10 Feb 2016 02:07:10 -0600
Subject: [Freeswitch-users] Hit that wallet now ! FreeSWITCH core lunch
 and dinner !
In-Reply-To: <CAPfjz3hzBQkyB42-BxV5xREa0U4ToXe-rtNDeicYS-itoh8ULw@mail.gmail.com>
References: <CALXCt0r3MA37Tv_GtULEysa3kQfOC_e-xYvNAYZJKpixs0tt5A@mail.gmail.com>
	<CA+Kn03=nx1C7WpsmAEgzXBSVWBR1q-KCFkw3KOSHA2YzSgyaTw@mail.gmail.com>
	<CAJ=fo8xvXcAS9PmEYDZ=pJ3rfp-8HNMZARaE-N7J6ndNLpgWZg@mail.gmail.com>
	<CAPfjz3hzBQkyB42-BxV5xREa0U4ToXe-rtNDeicYS-itoh8ULw@mail.gmail.com>
Message-ID: <CA+Kn03=L7x8C8kB6Ddn5cdNSVz0NFQtVOADzAdtMwLOSoarwBQ@mail.gmail.com>

+3

https://www.gofundme.com/freeswitch
On Feb 10, 2016 1:00 AM, "Jai Rangi" <jprangi at didforsale.com> wrote:

> +2
>
> *Jai Rangi*
> Cebod Technologies LLC dba DIDforSale/Cebod Telecom
> O 949-471-0102 <O%20949-471-0101> | C 949-419-7634 <C%20949-742-2666> | F
> 949-269-0449 / 949-232-1410 | jprangi at didforsale.com  www.cebod.com |
> www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA 92626  |
>
>
>
>
>
>
> On Tue, Feb 9, 2016 at 6:23 PM, Thomas Quintana <thomas at bettervoice.com>
> wrote:
>
>> +1
>>
>> Thomas Quintana
>> Chief Technology Officer
>> Phone: 1 (512) 677-6200
>> Website: http://www.bettervoice.com
>>
>>
>>
>> On Tue, Feb 9, 2016 at 1:26 PM, Blake Priddy <blake at cogents.io> wrote:
>>
>>> C'mon y'all!!!
>>>
>>> https://www.gofundme.com/freeswitch
>>> On Feb 9, 2016 12:22 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com>
>>> wrote:
>>>
>>>> Hey fellow FreeSWITCHers!
>>>>
>>>> How much we all made in 2015 thanks to FreeSWITCH?
>>>>
>>>> How much we made thanks to core devs (Anthony, Mike, Ken, Brian,
>>>> William, etc) answering our difficult questions?
>>>>
>>>> Let's send them MONEY so they can have a LUXURIOUS lunch and dinner on
>>>> us !
>>>>
>>>> On top right in http://www.freeswitch.org page there is a "Donate"
>>>> Paypal button that will directly translate in gluttony for the Core
>>>> FreeSWITCH Team.
>>>>
>>>> Core devs are having the annual development meeting days.
>>>>
>>>> If not now, when?
>>>>
>>>> Because!
>>>>
>>>> -giovanni
>>>>
>>>> --
>>>> 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
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 sdame at 207me.com  Wed Feb 10 16:20:17 2016
From: sdame at 207me.com (Stephen Dame)
Date: Wed, 10 Feb 2016 08:20:17 -0500
Subject: [Freeswitch-users] Hit that wallet now ! FreeSWITCH core lunch
	and dinner !
In-Reply-To: <CA+Kn03=L7x8C8kB6Ddn5cdNSVz0NFQtVOADzAdtMwLOSoarwBQ@mail.gmail.com>
References: <CALXCt0r3MA37Tv_GtULEysa3kQfOC_e-xYvNAYZJKpixs0tt5A@mail.gmail.com>	<CA+Kn03=nx1C7WpsmAEgzXBSVWBR1q-KCFkw3KOSHA2YzSgyaTw@mail.gmail.com>	<CAJ=fo8xvXcAS9PmEYDZ=pJ3rfp-8HNMZARaE-N7J6ndNLpgWZg@mail.gmail.com>	<CAPfjz3hzBQkyB42-BxV5xREa0U4ToXe-rtNDeicYS-itoh8ULw@mail.gmail.com>
	<CA+Kn03=L7x8C8kB6Ddn5cdNSVz0NFQtVOADzAdtMwLOSoarwBQ@mail.gmail.com>
Message-ID: <006501d16405$e4e71aa0$aeb54fe0$@207me.com>

+4

 

Regards,

Stephen

 

HostBBB ? Online Learning Solutions   <http://www.hostbbb.com/> http://www.hostbbb.com

207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Blake Priddy
Sent: Wednesday, February 10, 2016 3:07 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Hit that wallet now ! FreeSWITCH core lunch and dinner !

 

+3

https://www.gofundme.com/freeswitch

On Feb 10, 2016 1:00 AM, "Jai Rangi" <jprangi at didforsale.com <mailto:jprangi at didforsale.com> > wrote:

+2




Jai Rangi
Cebod Technologies LLC dba DIDforSale/Cebod Telecom 

 <tel:O%20949-471-0101> O 949-471-0102 |  <tel:C%20949-742-2666> C 949-419-7634 | F 949-269-0449 / 949-232-1410 |  <mailto:jprangi at didforsale.com> jprangi at didforsale.com   <http://www.cebod.com> www.cebod.com |  www.didforsale.com <http://www.didforsale.com>  |3200 Bristol St Suite 615, Costa Mesa, CA 92626  | 

  <https://docs.google.com/uc?export=download&id=0Bz0cNflKgNsWQno1V3dQTzhNSUk&revid=0Bz0cNflKgNsWOFh1ZnMydnBtU1ZDLzZ2OHBMM2Y2bVQ3R1pFPQ> 





 

On Tue, Feb 9, 2016 at 6:23 PM, Thomas Quintana <thomas at bettervoice.com <mailto:thomas at bettervoice.com> > wrote:

+1




Thomas Quintana

Chief Technology Officer

Phone: 1 (512) 677-6200 <tel:1%20%28512%29%20677-6200> 

Website: http://www.bettervoice.com

 

  <https://www.bettervoice.com/static/images/bvlogogrey.png> 

 

On Tue, Feb 9, 2016 at 1:26 PM, Blake Priddy <blake at cogents.io <mailto:blake at cogents.io> > wrote:

C'mon y'all!!!

https://www.gofundme.com/freeswitch

On Feb 9, 2016 12:22 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com> > wrote:

Hey fellow FreeSWITCHers!

How much we all made in 2015 thanks to FreeSWITCH?

How much we made thanks to core devs (Anthony, Mike, Ken, Brian, William, etc) answering our difficult questions?

Let's send them MONEY so they can have a LUXURIOUS lunch and dinner on us !

On top right in http://www.freeswitch.org page there is a "Donate" Paypal button that will directly translate in gluttony for the Core FreeSWITCH Team.

Core devs are having the annual development meeting days.

 

If not now, when?

 

Because!

-giovanni



-- 

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
http://lists.freeswitch.org/mailman/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 jjserranor at gmail.com  Wed Feb 10 13:31:56 2016
From: jjserranor at gmail.com (Jose Serrano)
Date: Wed, 10 Feb 2016 11:31:56 +0100
Subject: [Freeswitch-users] P-asserted-identity
Message-ID: <CAArQ-M=aa0rN-pXKe0OXfYy8bQs9qoNVS51TPopu37J0_e2XkA@mail.gmail.com>

Hello.

My freeswitch  by default replace the P-asserted-identity  by
Remote-Party-ID when routing calls to the gateway.
I want to send the P-asserted-identity and not the Remote-Party-ID and for
that I tried the following:

I have configured  in the outbound gateway definition the following
parameters:
<param name="caller-id-type" value="pid"/> or
<param name="cid-type" value="pid"/>
 but te behavior is the same.

The only thing that works is to set up the "sip_cid_type" variable in the
dialplan like this:
<action application="bridge"
data="{sip_cid_type=pid}sofia/gateway/Mygateway/$1"/>

Anyone  knows how I can send the P-Asserted-identity without having to
modify all my dial plan adding the sip_cid_type?

Thanks in avanced
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From emamirazavi at gmail.com  Wed Feb 10 17:18:52 2016
From: emamirazavi at gmail.com (S.Mohammad Emami Razavi)
Date: Wed, 10 Feb 2016 17:48:52 +0330
Subject: [Freeswitch-users] mod_verto and a1-hash
Message-ID: <CAJEstwPuHE_EKJ+vA_9cyHD3Ebc4Pr64SxQ45Hr20KBo0pHetA@mail.gmail.com>

Hello, I'm generating hashed password in the domain and with the username
in md5. But after setting it as a1-hash param in user directory, verto user
can not authenticate correctly and an error is returned from mod_verto.
After all authenticating with plain username and password and without
a1-hash set in user directory param, is excellent in mod_verto.
Can anybody help?!
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From v.kovalyshyn at gmail.com  Wed Feb 10 17:23:54 2016
From: v.kovalyshyn at gmail.com (Vitaly Kovalyshyn)
Date: Wed, 10 Feb 2016 16:23:54 +0200
Subject: [Freeswitch-users] mod_verto and a1-hash
In-Reply-To: <CAJEstwPuHE_EKJ+vA_9cyHD3Ebc4Pr64SxQ45Hr20KBo0pHetA@mail.gmail.com>
References: <CAJEstwPuHE_EKJ+vA_9cyHD3Ebc4Pr64SxQ45Hr20KBo0pHetA@mail.gmail.com>
Message-ID: <F31969F3-E696-497C-9671-2FDDD8B72181@gmail.com>

https://freeswitch.org/jira/browse/FS-6982 <https://freeswitch.org/jira/browse/FS-6982>


Best regards, 
Vitaly Kovalyshyn


> On 10 ???. 2016 ?., at 16:18, S.Mohammad Emami Razavi <emamirazavi at gmail.com> wrote:
> 
> Hello, I'm generating hashed password in the domain and with the username in md5. But after setting it as a1-hash param in user directory, verto user can not authenticate correctly and an error is returned from mod_verto. After all authenticating with plain username and password and without a1-hash set in user directory param, is excellent in mod_verto.
> Can anybody help?!
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From royj at yandex.ru  Wed Feb 10 17:33:43 2016
From: royj at yandex.ru (royj at yandex.ru)
Date: Wed, 10 Feb 2016 17:33:43 +0300
Subject: [Freeswitch-users] P-asserted-identity
In-Reply-To: <CAArQ-M=aa0rN-pXKe0OXfYy8bQs9qoNVS51TPopu37J0_e2XkA@mail.gmail.com>
References: <CAArQ-M=aa0rN-pXKe0OXfYy8bQs9qoNVS51TPopu37J0_e2XkA@mail.gmail.com>
Message-ID: <2566771455114823@web13h.yandex.ru>

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From mike at jerris.com  Wed Feb 10 18:54:29 2016
From: mike at jerris.com (Michael Jerris)
Date: Wed, 10 Feb 2016 09:54:29 -0600
Subject: [Freeswitch-users] Hit that wallet now ! FreeSWITCH core lunch
	and dinner !
In-Reply-To: <006501d16405$e4e71aa0$aeb54fe0$@207me.com>
References: <CALXCt0r3MA37Tv_GtULEysa3kQfOC_e-xYvNAYZJKpixs0tt5A@mail.gmail.com>
	<CA+Kn03=nx1C7WpsmAEgzXBSVWBR1q-KCFkw3KOSHA2YzSgyaTw@mail.gmail.com>
	<CAJ=fo8xvXcAS9PmEYDZ=pJ3rfp-8HNMZARaE-N7J6ndNLpgWZg@mail.gmail.com>
	<CAPfjz3hzBQkyB42-BxV5xREa0U4ToXe-rtNDeicYS-itoh8ULw@mail.gmail.com>
	<CA+Kn03=L7x8C8kB6Ddn5cdNSVz0NFQtVOADzAdtMwLOSoarwBQ@mail.gmail.com>
	<006501d16405$e4e71aa0$aeb54fe0$@207me.com>
Message-ID: <645A5847-61E2-42CE-94B4-EDF5AD7C6FD2@jerris.com>

while (x++) {
	printf("Thanks Everyone!!!\n");
	if (x == MAX_UINT64T) {
		printf("Whoa!!!\n");
	}
}

> On Feb 10, 2016, at 7:20 AM, Stephen Dame <sdame at 207me.com> wrote:
> 
> +4
>  
> Regards,
> Stephen
>  
> HostBBB ? Online Learning Solutions  http://www.hostbbb.com <http://www.hostbbb.com/>
> 207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Blake Priddy
> Sent: Wednesday, February 10, 2016 3:07 AM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Hit that wallet now ! FreeSWITCH core lunch and dinner !
>  
> +3
> 
> https://www.gofundme.com/freeswitch <https://www.gofundme.com/freeswitch>
> On Feb 10, 2016 1:00 AM, "Jai Rangi" <jprangi at didforsale.com <mailto:jprangi at didforsale.com>> wrote:
>> +2
>> 
>> Jai Rangi
>> Cebod Technologies LLC dba DIDforSale/Cebod Telecom 
>> O 949-471-0102 <tel:O%20949-471-0101> | C 949-419-7634 <tel:C%20949-742-2666> | F 949-269-0449 / 949-232-1410 | jprangi at didforsale.com <mailto:jprangi at didforsale.com>  www.cebod.com <http://www.cebod.com/> |  www.didforsale.com <http://www.didforsale.com/> |3200 Bristol St Suite 615, Costa Mesa, CA 92626  | 
>> 
>> 
>> 
>> 
>> 
>>  
>> On Tue, Feb 9, 2016 at 6:23 PM, Thomas Quintana <thomas at bettervoice.com <mailto:thomas at bettervoice.com>> wrote:
>>> +1
>>> 
>>> Thomas Quintana
>>> Chief Technology Officer
>>> Phone: 1 (512) 677-6200 <tel:1%20%28512%29%20677-6200>
>>> Website: http://www.bettervoice.com <http://www.bettervoice.com/>
>>>  
>>> 
>>>  
>>> On Tue, Feb 9, 2016 at 1:26 PM, Blake Priddy <blake at cogents.io <mailto:blake at cogents.io>> wrote:
>>>> C'mon y'all!!!
>>>> 
>>>> https://www.gofundme.com/freeswitch <https://www.gofundme.com/freeswitch>
>>>> On Feb 9, 2016 12:22 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>> wrote:
>>>>> Hey fellow FreeSWITCHers!
>>>>> 
>>>>> How much we all made in 2015 thanks to FreeSWITCH?
>>>>> 
>>>>> How much we made thanks to core devs (Anthony, Mike, Ken, Brian, William, etc) answering our difficult questions?
>>>>> 
>>>>> Let's send them MONEY so they can have a LUXURIOUS lunch and dinner on us !
>>>>> 
>>>>> On top right in http://www.freeswitch.org <http://www.freeswitch.org/> page there is a "Donate" Paypal button that will directly translate in gluttony for the Core FreeSWITCH Team.
>>>>> 
>>>>> Core devs are having the annual development meeting days.
>>>>>  
>>>>> If not now, when?
>>>>>  
>>>>> Because!
>>>>> 
>>>>> -giovanni
>>>>> 
>>>>> -- 
>>>>> 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 <http://www.freeswitchsolutions.com/>
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>>>>> http://www.cluecon.com <http://www.cluecon.com/>
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>>>> http://www.cluecon.com <http://www.cluecon.com/>
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>  
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>>> http://www.cluecon.com <http://www.cluecon.com/>
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>  
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>_________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb 10 18:56:47 2016
From: mike at jerris.com (Michael Jerris)
Date: Wed, 10 Feb 2016 09:56:47 -0600
Subject: [Freeswitch-users] P-asserted-identity
In-Reply-To: <CAArQ-M=aa0rN-pXKe0OXfYy8bQs9qoNVS51TPopu37J0_e2XkA@mail.gmail.com>
References: <CAArQ-M=aa0rN-pXKe0OXfYy8bQs9qoNVS51TPopu37J0_e2XkA@mail.gmail.com>
Message-ID: <D86648F3-3293-401F-9A1E-E84793E16274@jerris.com>

I think the other response answers your question, however I wanted to clarify one thing.  FreeSWITCH is a B2BUA.  It is not "replacing" the PAID with RPID, its a new call that is sent with its own configuration.

Mike

> On Feb 10, 2016, at 4:31 AM, Jose Serrano <jjserranor at gmail.com> wrote:
> 
> Hello.
> 
> My freeswitch  by default replace the P-asserted-identity  by Remote-Party-ID when routing calls to the gateway.
> I want to send the P-asserted-identity and not the Remote-Party-ID and for that I tried the following:
> 
> I have configured  in the outbound gateway definition the following parameters:
> <param name="caller-id-type" value="pid"/> or
> <param name="cid-type" value="pid"/>
>  but te behavior is the same.
> 
> The only thing that works is to set up the "sip_cid_type" variable in the dialplan like this:
> <action application="bridge" data="{sip_cid_type=pid}sofia/gateway/Mygateway/$1"/>
> 
> Anyone  knows how I can send the P-Asserted-identity without having to modify all my dial plan adding the sip_cid_type?
> 
> Thanks in avanced
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 nneul at mst.edu  Wed Feb 10 19:15:28 2016
From: nneul at mst.edu (Nathan Neulinger)
Date: Wed, 10 Feb 2016 10:15:28 -0600
Subject: [Freeswitch-users] Hit that wallet now ! FreeSWITCH core lunch
 and dinner !
In-Reply-To: <645A5847-61E2-42CE-94B4-EDF5AD7C6FD2@jerris.com>
References: <CALXCt0r3MA37Tv_GtULEysa3kQfOC_e-xYvNAYZJKpixs0tt5A@mail.gmail.com>
	<CA+Kn03=nx1C7WpsmAEgzXBSVWBR1q-KCFkw3KOSHA2YzSgyaTw@mail.gmail.com>
	<CAJ=fo8xvXcAS9PmEYDZ=pJ3rfp-8HNMZARaE-N7J6ndNLpgWZg@mail.gmail.com>
	<CAPfjz3hzBQkyB42-BxV5xREa0U4ToXe-rtNDeicYS-itoh8ULw@mail.gmail.com>
	<CA+Kn03=L7x8C8kB6Ddn5cdNSVz0NFQtVOADzAdtMwLOSoarwBQ@mail.gmail.com>
	<006501d16405$e4e71aa0$aeb54fe0$@207me.com>
	<645A5847-61E2-42CE-94B4-EDF5AD7C6FD2@jerris.com>
Message-ID: <56BB6220.3040605@mst.edu>

Better be careful there, that next loop takes all the money back.

:)

-- Nathan

On 02/10/2016 09:54 AM, Michael Jerris wrote:
> while (x++) {
> printf("Thanks Everyone!!!\n");
> if (x == MAX_UINT64T) {
> printf("Whoa!!!\n");
> }
> }
>
>> On Feb 10, 2016, at 7:20 AM, Stephen Dame <sdame at 207me.com <mailto:sdame at 207me.com>> wrote:
>>
>> +4
>> Regards,
>> Stephen
>> HostBBB ? Online Learning Solutions http://www.hostbbb.com <http://www.hostbbb.com/>
>> 207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame
>> *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*Blake Priddy
>> *Sent:*Wednesday, February 10, 2016 3:07 AM
>> *To:*FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
>> *Subject:*Re: [Freeswitch-users] Hit that wallet now ! FreeSWITCH core lunch and dinner !
>>
>> +3
>>
>> https://www.gofundme.com/freeswitch
>>
>> On Feb 10, 2016 1:00 AM, "Jai Rangi" <jprangi at didforsale.com <mailto:jprangi at didforsale.com>> wrote:
>>> +2
>>>
>>> *Jai Rangi*
>>> Cebod Technologies LLC dba DIDforSale/Cebod Telecom
>>> O 949-471-0102 <tel:O%20949-471-0101>|C 949-419-7634 <tel:C%20949-742-2666>|F 949-269-0449 /
>>> 949-232-1410|jprangi at didforsale.com <mailto:jprangi at didforsale.com> www.cebod.com
>>> <http://www.cebod.com/>|www.didforsale.com <http://www.didforsale.com/>|3200 Bristol St Suite 615, Costa Mesa, CA
>>> 92626  |
>>>
>>>
>>>
>>>
>>>
>>> On Tue, Feb 9, 2016 at 6:23 PM, Thomas Quintana <thomas at bettervoice.com <mailto:thomas at bettervoice.com>> wrote:
>>>> +1
>>>>
>>>> Thomas Quintana
>>>> Chief Technology Officer
>>>> Phone:1 (512) 677-6200 <tel:1%20%28512%29%20677-6200>
>>>> Website:http://www.bettervoice.com <http://www.bettervoice.com/>
>>>> On Tue, Feb 9, 2016 at 1:26 PM, Blake Priddy <blake at cogents.io <mailto:blake at cogents.io>> wrote:
>>>>>
>>>>> C'mon y'all!!!
>>>>>
>>>>> https://www.gofundme.com/freeswitch
>>>>>
>>>>> On Feb 9, 2016 12:22 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>> wrote:
>>>>>>
>>>>>> Hey fellow FreeSWITCHers!
>>>>>>
>>>>>> How much we all made in 2015 thanks to FreeSWITCH?
>>>>>>
>>>>>> How much we made thanks to core devs (Anthony, Mike, Ken, Brian, William, etc) answering our difficult questions?
>>>>>>
>>>>>> Let's send them MONEY so they can have a LUXURIOUS lunch and dinner on us !
>>>>>>
>>>>>> On top right inhttp://www.freeswitch.org <http://www.freeswitch.org/>page there is a "Donate" Paypal button that
>>>>>> will directly translate in gluttony for the Core FreeSWITCH Team.
>>>>>>
>>>>>> Core devs are having the annual development meeting days.
>>>>>> If not now, when?
>>>>>>
>>>>>> Because!
>>>>>>
>>>>>> -giovanni
>>>>>>
>>>>>> --
>>>>>> 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 <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 <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 <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://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 thomas.granvej6 at gmail.com  Wed Feb 10 19:48:22 2016
From: thomas.granvej6 at gmail.com (=?UTF-8?Q?Thomas_L=C3=B8cke?=)
Date: Wed, 10 Feb 2016 17:48:22 +0100
Subject: [Freeswitch-users] BEARERCAPABILITY_NOTAVAIL with 1.6.6 on Debian 8
Message-ID: <CAHJ5xP=HEd7Y5U1P4gOMPkAt2UVrn2VFB7CWhoype+UMQ8gCnA@mail.gmail.com>

Hi all,

Using 1.6.2 this works from fs_cli and dialplan:

originate loopback/park/default &bridge(sofia/gateway/server/xxxxxxxx)

Using 1.6.6 it fails with BEARERCAPABILITY_NOTAVAIL.

Has something changed between 1.6.2 and 1.6.6 that may be the cause of
this, and if so, what can I do to fix it?

:o)
Thomas
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From krice at freeswitch.org  Wed Feb 10 20:07:30 2016
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 10 Feb 2016 11:07:30 -0600
Subject: [Freeswitch-users] BEARERCAPABILITY_NOTAVAIL with 1.6.6 on
	Debian 8
In-Reply-To: <CAHJ5xP=HEd7Y5U1P4gOMPkAt2UVrn2VFB7CWhoype+UMQ8gCnA@mail.gmail.com>
References: <CAHJ5xP=HEd7Y5U1P4gOMPkAt2UVrn2VFB7CWhoype+UMQ8gCnA@mail.gmail.com>
Message-ID: <CAPcNkdetPA5WJsMTtUKDPVJ+JWf2zYLs7uafGZQgwV_u5_tNow@mail.gmail.com>

I doubt that was ever an intended feature... why not just send the A leg
out and park the bleg just reverse your notation there

On Wed, Feb 10, 2016 at 10:48 AM, Thomas L?cke <thomas.granvej6 at gmail.com>
wrote:

> Hi all,
>
> Using 1.6.2 this works from fs_cli and dialplan:
>
> originate loopback/park/default &bridge(sofia/gateway/server/xxxxxxxx)
>
> Using 1.6.6 it fails with BEARERCAPABILITY_NOTAVAIL.
>
> Has something changed between 1.6.2 and 1.6.6 that may be the cause of
> this, and if so, what can I do to fix it?
>
> :o)
> Thomas
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 thomas.granvej6 at gmail.com  Wed Feb 10 20:16:19 2016
From: thomas.granvej6 at gmail.com (=?UTF-8?Q?Thomas_L=C3=B8cke?=)
Date: Wed, 10 Feb 2016 18:16:19 +0100
Subject: [Freeswitch-users] BEARERCAPABILITY_NOTAVAIL with 1.6.6 on
	Debian 8
In-Reply-To: <CAPcNkdetPA5WJsMTtUKDPVJ+JWf2zYLs7uafGZQgwV_u5_tNow@mail.gmail.com>
References: <CAHJ5xP=HEd7Y5U1P4gOMPkAt2UVrn2VFB7CWhoype+UMQ8gCnA@mail.gmail.com>
	<CAPcNkdetPA5WJsMTtUKDPVJ+JWf2zYLs7uafGZQgwV_u5_tNow@mail.gmail.com>
Message-ID: <CAHJ5xPmcOOPdfphZ-NQS2eZstP5F+8PqTykj=e9wfu1qWmWcVw@mail.gmail.com>

Hi Ken,

It doesn't work from our dialplan either:

<extension name="foobar" continue="true">
      <condition field="destination_number"
expression="^external_transfer_(\d+)$">
        <action application="set" data="ringback=${dk-ring}"/>
        <action application="ring_ready" />
        <action application="bridge"
data="[fifo_music=default]sofia/gateway/${default_trunk}/$1"/>
        <action application="hangup"/>
      </condition>
    </extension>

That works swimmingly with 1.6.2, but fails with BEARERCAPABILITY_NOTAVAIL
with 1.6.6.


2016-02-10 18:07 GMT+01:00 Ken Rice <krice at freeswitch.org>:

> I doubt that was ever an intended feature... why not just send the A leg
> out and park the bleg just reverse your notation there
>
> On Wed, Feb 10, 2016 at 10:48 AM, Thomas L?cke <thomas.granvej6 at gmail.com>
> wrote:
>
>> Hi all,
>>
>> Using 1.6.2 this works from fs_cli and dialplan:
>>
>> originate loopback/park/default &bridge(sofia/gateway/server/xxxxxxxx)
>>
>> Using 1.6.6 it fails with BEARERCAPABILITY_NOTAVAIL.
>>
>
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From brian at freeswitch.org  Wed Feb 10 20:18:42 2016
From: brian at freeswitch.org (Brian West)
Date: Wed, 10 Feb 2016 11:18:42 -0600
Subject: [Freeswitch-users] BEARERCAPABILITY_NOTAVAIL with 1.6.6 on
	Debian 8
In-Reply-To: <CAHJ5xPmcOOPdfphZ-NQS2eZstP5F+8PqTykj=e9wfu1qWmWcVw@mail.gmail.com>
References: <CAHJ5xP=HEd7Y5U1P4gOMPkAt2UVrn2VFB7CWhoype+UMQ8gCnA@mail.gmail.com>
	<CAPcNkdetPA5WJsMTtUKDPVJ+JWf2zYLs7uafGZQgwV_u5_tNow@mail.gmail.com>
	<CAHJ5xPmcOOPdfphZ-NQS2eZstP5F+8PqTykj=e9wfu1qWmWcVw@mail.gmail.com>
Message-ID: <CAEJMVkArxF9v30Q_icMjWzMiuCYbb_oXPWpuCb4S0i8sTLDgdg@mail.gmail.com>

Look at the sip signaling

On Wednesday, February 10, 2016, Thomas L?cke <thomas.granvej6 at gmail.com>
wrote:

> Hi Ken,
>
> It doesn't work from our dialplan either:
>
> <extension name="foobar" continue="true">
>       <condition field="destination_number"
> expression="^external_transfer_(\d+)$">
>         <action application="set" data="ringback=${dk-ring}"/>
>         <action application="ring_ready" />
>         <action application="bridge"
> data="[fifo_music=default]sofia/gateway/${default_trunk}/$1"/>
>         <action application="hangup"/>
>       </condition>
>     </extension>
>
> That works swimmingly with 1.6.2, but fails with BEARERCAPABILITY_NOTAVAIL
> with 1.6.6.
>
>
> 2016-02-10 18:07 GMT+01:00 Ken Rice <krice at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','krice at freeswitch.org');>>:
>
>> I doubt that was ever an intended feature... why not just send the A leg
>> out and park the bleg just reverse your notation there
>>
>> On Wed, Feb 10, 2016 at 10:48 AM, Thomas L?cke <thomas.granvej6 at gmail.com
>> <javascript:_e(%7B%7D,'cvml','thomas.granvej6 at gmail.com');>> wrote:
>>
>>> Hi all,
>>>
>>> Using 1.6.2 this works from fs_cli and dialplan:
>>>
>>> originate loopback/park/default &bridge(sofia/gateway/server/xxxxxxxx)
>>>
>>> Using 1.6.6 it fails with BEARERCAPABILITY_NOTAVAIL.
>>>
>>

-- 

*Brian West*
brian at freeswitch.org


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

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 jjserranor at gmail.com  Wed Feb 10 22:38:42 2016
From: jjserranor at gmail.com (Jose Serrano)
Date: Wed, 10 Feb 2016 20:38:42 +0100
Subject: [Freeswitch-users] P-asserted-identity
In-Reply-To: <D86648F3-3293-401F-9A1E-E84793E16274@jerris.com>
References: <CAArQ-M=aa0rN-pXKe0OXfYy8bQs9qoNVS51TPopu37J0_e2XkA@mail.gmail.com>
	<D86648F3-3293-401F-9A1E-E84793E16274@jerris.com>
Message-ID: <CAArQ-Mk3ocjWQgXh3AycEffjVVY3pWu2oAUfQwQDuF4mdT_6Pg@mail.gmail.com>

Thanks both for your answer.

However
I tried to do what royj suggested me and still the same behaviour (
freeswith send Remote-Party-ID and not P-asserted-id)
I have found a old bug that match exactly with my issue:
https://freeswitch.org/jira/browse/FS-527.
Yes!, it is closed, but is reproducing to me with my current version of
freeswitch 1.4.20

what do you think about it?


2016-02-10 16:56 GMT+01:00 Michael Jerris <mike at jerris.com>:

> I think the other response answers your question, however I wanted to
> clarify one thing.  FreeSWITCH is a B2BUA.  It is not "replacing" the PAID
> with RPID, its a new call that is sent with its own configuration.
>
> Mike
>
> > On Feb 10, 2016, at 4:31 AM, Jose Serrano <jjserranor at gmail.com> wrote:
> >
> > Hello.
> >
> > My freeswitch  by default replace the P-asserted-identity  by
> Remote-Party-ID when routing calls to the gateway.
> > I want to send the P-asserted-identity and not the Remote-Party-ID and
> for that I tried the following:
> >
> > I have configured  in the outbound gateway definition the following
> parameters:
> > <param name="caller-id-type" value="pid"/> or
> > <param name="cid-type" value="pid"/>
> >  but te behavior is the same.
> >
> > The only thing that works is to set up the "sip_cid_type" variable in
> the dialplan like this:
> > <action application="bridge"
> data="{sip_cid_type=pid}sofia/gateway/Mygateway/$1"/>
> >
> > Anyone  knows how I can send the P-Asserted-identity without having to
> modify all my dial plan adding the sip_cid_type?
> >
> > Thanks in avanced
> >
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 shafeeq.v at gmail.com  Wed Feb 10 23:59:07 2016
From: shafeeq.v at gmail.com (mohammed shafeeque)
Date: Thu, 11 Feb 2016 02:29:07 +0530
Subject: [Freeswitch-users] Oneway audio issues in freeswitch
Message-ID: <CANw+5nWmd_j=eUz7OGoH=5Wb3g43VOS_XF_dZVkuFRMRUtfZ2Q@mail.gmail.com>

Hello All

We are getting one way audio issues with some softphones and grandstream
phones behind nat registerd to our freeswitch server.

Here is scenario:
Grandstream call any extensions (one way audio)
Any extension call Grandstream ( Audio works just fine)

We have tried multiple softphones and the result is same.

Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF
issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started
with an upgrade to freeswitch.

Any help or hint will be much appreciated.

Thank you,
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From italo at freeswitch.org  Thu Feb 11 04:54:06 2016
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Wed, 10 Feb 2016 22:54:06 -0300
Subject: [Freeswitch-users] Oneway audio issues in freeswitch
In-Reply-To: <CANw+5nWmd_j=eUz7OGoH=5Wb3g43VOS_XF_dZVkuFRMRUtfZ2Q@mail.gmail.com>
References: <CANw+5nWmd_j=eUz7OGoH=5Wb3g43VOS_XF_dZVkuFRMRUtfZ2Q@mail.gmail.com>
Message-ID: <CABMX_w_NWvQ5ZUdxtYHnua4FD9i4hYDt3YeP-X+43XYi4cb=8A@mail.gmail.com>

You need to look at the sip signaling to see what's going on

On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque <shafeeq.v at gmail.com>
wrote:

> Hello All
>
> We are getting one way audio issues with some softphones and grandstream
> phones behind nat registerd to our freeswitch server.
>
> Here is scenario:
> Grandstream call any extensions (one way audio)
> Any extension call Grandstream ( Audio works just fine)
>
> We have tried multiple softphones and the result is same.
>
> Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF
> issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started
> with an upgrade to freeswitch.
>
> Any help or hint will be much appreciated.
>
> Thank you,
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
?talo Rossi
italo at freeswitch.org
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From jprangi at gmail.com  Thu Feb 11 06:45:30 2016
From: jprangi at gmail.com (Jai Rangi)
Date: Wed, 10 Feb 2016 19:45:30 -0800
Subject: [Freeswitch-users] Oneway audio issues in freeswitch
In-Reply-To: <CABMX_w_NWvQ5ZUdxtYHnua4FD9i4hYDt3YeP-X+43XYi4cb=8A@mail.gmail.com>
References: <CANw+5nWmd_j=eUz7OGoH=5Wb3g43VOS_XF_dZVkuFRMRUtfZ2Q@mail.gmail.com>
	<CABMX_w_NWvQ5ZUdxtYHnua4FD9i4hYDt3YeP-X+43XYi4cb=8A@mail.gmail.com>
Message-ID: <CAPfjz3hD0P5eeXjY_QQrh863MLmdCk09boOFFB5T+=WjCAB6sA@mail.gmail.com>

We have been looking at that all day, but cant figure out the issue. Funny
thing is that its happening only when GS Originate the call. May be we are
over looking something. Here are two call example. IPs are modified for
security.

FreeSWITCH (Version 1.6.6  64bit) is ready
freeswitch at internal> sofia_contact 1276 at domain.company.com
sofia/internal/sip:1276 at 192.168.1.168:12113
;fs_nat=yes;fs_path=sip%3A1276%4068.5.194.163%3A12113
freeswitch at internal> sofia_contact 142 at domain.company.com
sofia/internal/sip:142 at 172.16.42.13:11852
;fs_nat=yes;fs_path=sip%3A142%4074.67.200.39%3A33812


142 calls 1276 (1276 does not hear anything) (Seems freeswitch not handling
nat properly) On TCP dump, I can see free switching receiving the RTP
packet, but trying to deliver to local IP for 1276.
1276 calls 142 (All good both parties can hear)
142 call PSTN number (All good)
1272 call PSTN number (All good)

Same configuration, same dialplan works just fine with 1.6.2 and 1.4.18.
1.6.2 had intermittent DTMF issue, we upgraded to 1.6.5, found this one way
audio, upgraded to 1.6.6.
We have narrowed it down to Grand Stream Softphone and Grad Stream IP
phones.
Here is trace for Both calls.

######################################
U 74.67.200.39:33812 -> 222.222.222.222:5060
INVITE sip:1276 at domain.company.com:5060 SIP/2.0.
Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK1892445982;rport.
From: "142" <sip:142 at domain.company.com:5060>;tag=479799221.
To: <sip:1276 at domain.company.com:5060>.
Call-ID: 152039027-11852-15 at BHC.BG.EC.BD.
CSeq: 140 INVITE.
Contact: "142" <sip:142 at 172.16.42.13:11852>.
Max-Forwards: 70.
User-Agent: Grandstream Wave/IOS 1.0.19.
Privacy: none.
P-Preferred-Identity: "142" <sip:142 at domain.company.com:5060>.
Supported: replaces, path, timer, eventlist.
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE, MESSAGE.
Content-Type: application/sdp.
Accept: application/sdp, application/dtmf-relay.
Content-Length:   235.
.
v=0.
o=142 8000 8000 IN IP4 172.16.42.13.
s=SIP Call.
c=IN IP4 172.16.42.13.
t=0 0.
m=audio 50476 RTP/AVP 0 8 101.
a=sendrecv.
a=rtpmap:0 PCMU/8000.
a=ptime:20.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.

#
U 222.222.222.222:5060 -> 74.67.200.39:33812
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 172.16.42.13:11852
;branch=z9hG4bK1892445982;rport=33812;received=74.67.200.39.
From: "142" <sip:142 at domain.company.com:5060>;tag=479799221.
To: <sip:1276 at domain.company.com:5060>.
Call-ID: 152039027-11852-15 at BHC.BG.EC.BD.
CSeq: 140 INVITE.
User-Agent: VOIPGATEWAY.
Content-Length: 0.
.

#
U 222.222.222.222:5060 -> 74.67.200.39:33812
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP 172.16.42.13:11852
;branch=z9hG4bK1892445982;rport=33812;received=74.67.200.39.
From: "142" <sip:142 at domain.company.com:5060>;tag=479799221.
To: <sip:1276 at domain.company.com:5060>;tag=Kt2jU0QN57N8F.
Call-ID: 152039027-11852-15 at BHC.BG.EC.BD.
CSeq: 140 INVITE.
User-Agent: VOIPGATEWAY.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: path, replaces.
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer.
Proxy-Authenticate: Digest realm="domain.company.com",
nonce="5eb04922-17e3-426e-9ddf-4333692102b5", algorithm=MD5, qop="auth".
Content-Length: 0.
.

##
U 74.67.200.39:33812 -> 222.222.222.222:5060
ACK sip:1276 at domain.company.com:5060 SIP/2.0.
Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK1892445982;rport.
From: "142" <sip:142 at domain.company.com:5060>;tag=479799221.
To: <sip:1276 at domain.company.com:5060>;tag=Kt2jU0QN57N8F.
Call-ID: 152039027-11852-15 at BHC.BG.EC.BD.
CSeq: 140 ACK.
Content-Length: 0.
.

#
U 74.67.200.39:33812 -> 222.222.222.222:5060
INVITE sip:1276 at domain.company.com:5060 SIP/2.0.
Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK322043518;rport.
From: "142" <sip:142 at domain.company.com:5060>;tag=479799221.
To: <sip:1276 at domain.company.com:5060>.
Call-ID: 152039027-11852-15 at BHC.BG.EC.BD.
CSeq: 141 INVITE.
Contact: "142" <sip:142 at 172.16.42.13:11852>.
Proxy-Authorization: Digest username="xxxxx", realm="domain.company.com",
nonce="5eb04922-17e3-426e-9ddf-4333692102b5", uri="
sip:1276 at domain.company.com:5060",
response="d309cb76b83042023f6794835ad60a89", algorithm=MD5,
cnonce="15121380", qop=auth, nc=00000003.
Max-Forwards: 70.
User-Agent: Grandstream Wave/IOS 1.0.19.
Privacy: none.
P-Preferred-Identity: "142" <sip:142 at domain.company.com:5060>.
Supported: replaces, path, timer, eventlist.
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE, MESSAGE.
Content-Type: application/sdp.
Accept: application/sdp, application/dtmf-relay.
Content-Length:   235.
.
v=0.
o=142 8000 8000 IN IP4 172.16.42.13.
s=SIP Call.
c=IN IP4 172.16.42.13.
t=0 0.
m=audio 50476 RTP/AVP 0 8 101.
a=sendrecv.
a=rtpmap:0 PCMU/8000.
a=ptime:20.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.

#
U 222.222.222.222:5060 -> 74.67.200.39:33812
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 172.16.42.13:11852
;branch=z9hG4bK322043518;rport=33812;received=74.67.200.39.
From: "142" <sip:142 at domain.company.com:5060>;tag=479799221.
To: <sip:1276 at domain.company.com:5060>.
Call-ID: 152039027-11852-15 at BHC.BG.EC.BD.
CSeq: 141 INVITE.
User-Agent: VOIPGATEWAY.
Content-Length: 0.
.

#
U 222.222.222.222:5060 -> 68.5.194.163:12113
INVITE sip:1276 at 192.168.1.168:12113 SIP/2.0.
Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bKX5H9FDZX1eyNN.
Route: <sip:1276 at 68.5.194.163:12113>.
Max-Forwards: 68.
From: "Softphone" <sip:142 at domain.company.com>;tag=NcN4ypSvZS2DQ.
To: <sip:1276 at 192.168.1.168:12113>.
Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243919 INVITE.
Contact: <sip:142 at 222.222.222.222:5060>.
User-Agent: VOIPGATEWAY.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: path, replaces.
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 270.
X-FS-Support: update_display,send_info.
.
v=0.
o=FreeSWITCH 1455142324 1455142325 IN IP4 222.222.222.222.
s=FreeSWITCH.
c=IN IP4 222.222.222.222.
t=0 0.
m=audio 17642 RTP/AVP 0 8 101 13.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.

#
#
U 68.5.194.163:12113 -> 222.222.222.222:5060
SIP/2.0 100 Trying.
From: Softphone<sip:142 at domain.company.com>;tag=NcN4ypSvZS2DQ.
To: sip:1276 at 192.168.1.168:12113.
Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243919 INVITE.
Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bKX5H9FDZX1eyNN.
Content-Length: 0.
.

###
U 68.5.194.163:12113 -> 222.222.222.222:5060
SIP/2.0 180 Ringing.
From: Softphone<sip:142 at domain.company.com>;tag=NcN4ypSvZS2DQ.
To: sip:1276 at 192.168.1.168:12113;tag=ReKf2-xIWCt.
Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243919 INVITE.
Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bKX5H9FDZX1eyNN.
Contact: ATAPHONE<sip:1276 at 192.168.1.168:12113>.
User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>.
Content-Length: 0.
.

#
U 222.222.222.222:5060 -> 74.67.200.39:33812
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP 172.16.42.13:11852
;branch=z9hG4bK322043518;rport=33812;received=74.67.200.39.
From: "142" <sip:142 at domain.company.com:5060>;tag=479799221.
To: <sip:1276 at domain.company.com:5060>;tag=m3UBXU8r2gcUB.
Call-ID: 152039027-11852-15 at BHC.BG.EC.BD.
CSeq: 141 INVITE.
Contact: <sip:1276 at 222.222.222.222:5060;transport=udp>.
User-Agent: VOIPGATEWAY.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: path, replaces.
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 222.
P-Asserted-Identity: "1276" <sip:1276 at domain.company.com>.
.
v=0.
o=FreeSWITCH 1455132056 1455132057 IN IP4 222.222.222.222.
s=FreeSWITCH.
c=IN IP4 222.222.222.222.
t=0 0.
m=audio 27910 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.

##############
##
#######
U 68.5.194.163:12113 -> 222.222.222.222:5060
SIP/2.0 200 OK.
From: Softphone<sip:142 at domain.company.com>;tag=NcN4ypSvZS2DQ.
To: sip:1276 at 192.168.1.168:12113;tag=ReKf2-xIWCt.
Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243919 INVITE.
Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bKX5H9FDZX1eyNN.
Contact: ATAPHONE<sip:1276 at 192.168.1.168:12113>.
User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>.
Allow: INVITE,BYE,CANCEL,OPTIONS,PRACK,NOTIFY,UPDATE,REFER.
Supported: timer,replaces.
Content-Type: application/sdp.
Content-Length: 207.
.
v=0.
o=1276 87748 1 IN IP4 192.168.1.168.
s=-.
c=IN IP4 192.168.1.168.
t=0 0.
m=audio 8000 RTP/AVP 0 96.
a=rtpmap:0 PCMU/8000.
a=rtpmap:96 telephone-event/8000.
a=ptime:20.
a=rtpmap:96 telephone-event/8000.

#
U 222.222.222.222:5060 -> 68.5.194.163:12113
ACK sip:1276 at 192.168.1.168:12113 SIP/2.0.
Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bKZQ4tK304U0aUc.
Max-Forwards: 70.
From: "Softphone" <sip:142 at domain.company.com>;tag=NcN4ypSvZS2DQ.
To: <sip:1276 at 192.168.1.168:12113>;tag=ReKf2-xIWCt.
Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243919 ACK.
Contact: <sip:142 at 222.222.222.222:5060>.
Content-Length: 0.
.

#
U 222.222.222.222:5060 -> 74.67.200.39:33812
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 172.16.42.13:11852
;branch=z9hG4bK322043518;rport=33812;received=74.67.200.39.
From: "142" <sip:142 at domain.company.com:5060>;tag=479799221.
To: <sip:1276 at domain.company.com:5060>;tag=m3UBXU8r2gcUB.
Call-ID: 152039027-11852-15 at BHC.BG.EC.BD.
CSeq: 141 INVITE.
Contact: <sip:1276 at 222.222.222.222:5060;transport=udp>.
User-Agent: VOIPGATEWAY.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: path, replaces.
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 222.
P-Asserted-Identity: "Outbound Call" <sip:1276 at domain.company.com>.
.
v=0.
o=FreeSWITCH 1455132056 1455132057 IN IP4 222.222.222.222.
s=FreeSWITCH.
c=IN IP4 222.222.222.222.
t=0 0.
m=audio 27910 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.

##
U 74.67.200.39:33812 -> 222.222.222.222:5060
ACK sip:1276 at 222.222.222.222:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK1657841580;rport.
From: "142" <sip:142 at domain.company.com:5060>;tag=479799221.
To: <sip:1276 at domain.company.com:5060>;tag=m3UBXU8r2gcUB.
Call-ID: 152039027-11852-15 at BHC.BG.EC.BD.
CSeq: 141 ACK.
Contact: <sip:142 at 172.16.42.13:11852>.
Proxy-Authorization: Digest username="xxxx", realm="domain.company.com",
nonce="5eb04922-17e3-426e-9ddf-4333692102b5", uri="
sip:1276 at domain.company.com:5060",
response="d309cb76b83042023f6794835ad60a89", algorithm=MD5,
cnonce="15121380", qop=auth, nc=00000003.
Max-Forwards: 70.
Supported: replaces, path, timer, eventlist.
User-Agent: Grandstream Wave/IOS 1.0.19.
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE, MESSAGE.
Content-Length: 0.
.

###############################################################
####
#

#
#########
################################################################################################
U 74.67.200.39:33812 -> 222.222.222.222:5060
BYE sip:1276 at 222.222.222.222:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK326133779;rport.
From: "142" <sip:142 at domain.company.com:5060>;tag=479799221.
To: <sip:1276 at domain.company.com:5060>;tag=m3UBXU8r2gcUB.
Call-ID: 152039027-11852-15 at BHC.BG.EC.BD.
CSeq: 142 BYE.
Contact: <sip:142 at 172.16.42.13:11852>.
Max-Forwards: 70.
Supported: replaces, path, timer, eventlist.
User-Agent: Grandstream Wave/IOS 1.0.19.
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE, MESSAGE.
Content-Length: 0.
.

#
U 222.222.222.222:5060 -> 74.67.200.39:33812
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 172.16.42.13:11852
;branch=z9hG4bK326133779;rport=33812;received=74.67.200.39.
From: "142" <sip:142 at domain.company.com:5060>;tag=479799221.
To: <sip:1276 at domain.company.com:5060>;tag=m3UBXU8r2gcUB.
Call-ID: 152039027-11852-15 at BHC.BG.EC.BD.
CSeq: 142 BYE.
User-Agent: VOIPGATEWAY.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: path, replaces.
Content-Length: 0.
.

###
U 222.222.222.222:5060 -> 68.5.194.163:12113
BYE sip:1276 at 192.168.1.168:12113 SIP/2.0.
Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bK2jg5rmKFKUDKF.
Max-Forwards: 70.
From: "Softphone" <sip:142 at domain.company.com>;tag=NcN4ypSvZS2DQ.
To: <sip:1276 at 192.168.1.168:12113>;tag=ReKf2-xIWCt.
Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243920 BYE.
User-Agent: VOIPGATEWAY.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: path, replaces.
Reason: Q.850;cause=16;text="NORMAL_CLEARING".
Content-Length: 0.
.

###
U 68.5.194.163:12113 -> 222.222.222.222:5060
SIP/2.0 200 OK.
From: Softphone<sip:142 at domain.company.com>;tag=NcN4ypSvZS2DQ.
To: sip:1276 at 192.168.1.168:12113;tag=ReKf2-xIWCt.
Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243920 BYE.
Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bK2jg5rmKFKUDKF.
User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>.
Content-Length: 0.
.

#######################################################
#########
#

Successful Call with 2 way audio :1276 calls 142

#
#
#
##################################
U 68.5.194.163:12113 -> 222.222.222.222:5060
INVITE sip:142 at domain.company.com SIP/2.0.
From: ATAPHONE<sip:1276 at domain.company.com>;tag=VxLf2-dmKe50.
To: sip:142 at domain.company.com.
Call-ID: d44p80-lYfd5f2 at domain.company.com.
CSeq: 176 INVITE.
Via: SIP/2.0/UDP 192.168.1.168:12113;branch=z9hG4bKu22f2-74MJFSKH.
Contact: ATAPHONE<sip:1276 at 192.168.1.168:12113>.
Max-Forwards: 70.
User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>.
Allow: INVITE,BYE,CANCEL,OPTIONS,PRACK,NOTIFY,UPDATE,REFER.
Supported: timer,replaces.
Content-Type: application/sdp.
Content-Length: 257.
.
v=0.
o=1276 87384 1 IN IP4 192.168.1.168.
s=-.
c=IN IP4 192.168.1.168.
t=0 0.
m=audio 8002 RTP/AVP 0 18 8 96.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:96 telephone-event/8000.
a=ptime:20.
a=rtpmap:96 telephone-event/8000.

#
U 222.222.222.222:5060 -> 68.5.194.163:12113
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 192.168.1.168:12113
;branch=z9hG4bKu22f2-74MJFSKH;received=68.5.194.163.
From: ATAPHONE <sip:1276 at domain.company.com>;tag=VxLf2-dmKe50.
To: <sip:142 at domain.company.com>.
Call-ID: d44p80-lYfd5f2 at domain.company.com.
CSeq: 176 INVITE.
User-Agent: VOIPGATEWAY.
Content-Length: 0.
.

#
U 222.222.222.222:5060 -> 68.5.194.163:12113
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP 192.168.1.168:12113
;branch=z9hG4bKu22f2-74MJFSKH;received=68.5.194.163.
From: ATAPHONE <sip:1276 at domain.company.com>;tag=VxLf2-dmKe50.
To: <sip:142 at domain.company.com>;tag=SjctQ6jcUQD5a.
Call-ID: d44p80-lYfd5f2 at domain.company.com.
CSeq: 176 INVITE.
User-Agent: VOIPGATEWAY.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: path, replaces.
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer.
Proxy-Authenticate: Digest realm="domain.company.com",
nonce="1cf240ca-c5d1-4ac0-8417-becd41784ffb", algorithm=MD5, qop="auth".
Content-Length: 0.
.

#
U 68.5.194.163:12113 -> 222.222.222.222:5060
ACK sip:142 at domain.company.com SIP/2.0.
From: ATAPHONE<sip:1276 at domain.company.com>;tag=VxLf2-dmKe50.
To: sip:142 at domain.company.com;tag=SjctQ6jcUQD5a.
Call-ID: d44p80-lYfd5f2 at domain.company.com.
CSeq: 176 ACK.
Via: SIP/2.0/UDP 192.168.1.168:12113;branch=z9hG4bKu22f2-74MJFSKH.
Content-Length: 0.
.

#
U 68.5.194.163:12113 -> 222.222.222.222:5060
INVITE sip:142 at domain.company.com SIP/2.0.
From: ATAPHONE<sip:1276 at domain.company.com>;tag=VxLf2-dmKe50.
To: sip:142 at domain.company.com.
Call-ID: d44p80-lYfd5f2 at domain.company.com.
CSeq: 177 INVITE.
Via: SIP/2.0/UDP 192.168.1.168:12113;branch=z9hG4bKu22f2-l5MoOcq0.
Contact: ATAPHONE<sip:1276 at 192.168.1.168:12113>.
Max-Forwards: 70.
User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>.
Allow: INVITE,BYE,CANCEL,OPTIONS,PRACK,NOTIFY,UPDATE,REFER.
Supported: timer,replaces.
Proxy-Authorization: Digest username="yyyy",realm="domain.company.com",uri="
sip:142 at domain.company.com
",response="a87ca168b1869994a6b4782df7bffe99",algorithm=MD5,nonce="1cf240ca-c5d1-4ac0-8417-becd41784ffb",qop=auth,cnonce="00017214",nc=00000001.
Content-Type: application/sdp.
Content-Length: 257.
.
v=0.
o=1276 87384 1 IN IP4 192.168.1.168.
s=-.
c=IN IP4 192.168.1.168.
t=0 0.
m=audio 8002 RTP/AVP 0 18 8 96.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:96 telephone-event/8000.
a=ptime:20.
a=rtpmap:96 telephone-event/8000.

#
U 222.222.222.222:5060 -> 68.5.194.163:12113
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 192.168.1.168:12113
;branch=z9hG4bKu22f2-l5MoOcq0;received=68.5.194.163.
From: ATAPHONE <sip:1276 at domain.company.com>;tag=VxLf2-dmKe50.
To: <sip:142 at domain.company.com>.
Call-ID: d44p80-lYfd5f2 at domain.company.com.
CSeq: 177 INVITE.
User-Agent: VOIPGATEWAY.
Content-Length: 0.
.

#
U 222.222.222.222:5060 -> 74.67.200.39:33812
INVITE sip:142 at 172.16.42.13:11852 SIP/2.0.
Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bK6pN8Z0pX7y6XD.
Route: <sip:142 at 74.67.200.39:33812>.
Max-Forwards: 68.
From: "ATA" <sip:1276 at domain.company.com>;tag=U4yBUvmKN9Saj.
To: <sip:142 at 172.16.42.13:11852>.
Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243932 INVITE.
Contact: <sip:1276 at 222.222.222.222:5060>.
User-Agent: VOIPGATEWAY.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: path, replaces.
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 270.
X-FS-Support: update_display,send_info.
.
v=0.
o=FreeSWITCH 1455141647 1455141648 IN IP4 222.222.222.222.
s=FreeSWITCH.
c=IN IP4 222.222.222.222.
t=0 0.
m=audio 18346 RTP/AVP 0 8 101 13.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.

#

U 222.222.222.222:5060 -> 74.67.200.39:33812
INVITE sip:142 at 172.16.42.13:11852 SIP/2.0.
Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bK6pN8Z0pX7y6XD.
Route: <sip:142 at 74.67.200.39:33812>.
Max-Forwards: 68.
From: "ATA" <sip:1276 at domain.company.com>;tag=U4yBUvmKN9Saj.
To: <sip:142 at 172.16.42.13:11852>.
Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243932 INVITE.
Contact: <sip:1276 at 222.222.222.222:5060>.
User-Agent: VOIPGATEWAY.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: path, replaces.
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 270.
X-FS-Support: update_display,send_info.
.
v=0.
o=FreeSWITCH 1455141647 1455141648 IN IP4 222.222.222.222.
s=FreeSWITCH.
c=IN IP4 222.222.222.222.
t=0 0.
m=audio 18346 RTP/AVP 0 8 101 13.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.

##
U 74.67.200.39:33812 -> 222.222.222.222:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 222.222.222.222:5060
;rport=5660;branch=z9hG4bK6pN8Z0pX7y6XD.
From: "ATA" <sip:1276 at domain.company.com>;tag=U4yBUvmKN9Saj.
To: <sip:142 at 172.16.42.13:11852>.
Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243932 INVITE.
Supported: replaces, path, eventlist.
User-Agent: Grandstream Wave/IOS 1.0.19.
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE, MESSAGE.
Content-Length: 0.
.

#
U 74.67.200.39:33812 -> 222.222.222.222:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 222.222.222.222:5060
;rport=5660;branch=z9hG4bK6pN8Z0pX7y6XD.
From: "ATA" <sip:1276 at domain.company.com>;tag=U4yBUvmKN9Saj.
To: <sip:142 at 172.16.42.13:11852>;tag=1713531477.
Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243932 INVITE.
Contact: <sip:142 at 172.16.42.13:11852>.
Supported: replaces, path, timer, eventlist.
User-Agent: Grandstream Wave/IOS 1.0.19.
Allow-Events: talk, hold.
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE, MESSAGE.
Content-Length: 0.
.

#
U 222.222.222.222:5060 -> 68.5.194.163:12113
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP 192.168.1.168:12113
;branch=z9hG4bKu22f2-l5MoOcq0;received=68.5.194.163.
From: ATAPHONE <sip:1276 at domain.company.com>;tag=VxLf2-dmKe50.
To: <sip:142 at domain.company.com>;tag=tU5jS13Fr03Qp.
Call-ID: d44p80-lYfd5f2 at domain.company.com.
CSeq: 177 INVITE.
Contact: <sip:142 at 222.222.222.222:5060;transport=udp>.
User-Agent: VOIPGATEWAY.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: path, replaces.
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 219.
Remote-Party-ID: "142" <sip:142 at domain.company.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1455140898 1455140899 IN IP4 222.222.222.222.
s=FreeSWITCH.
c=IN IP4 222.222.222.222.
t=0 0.
m=audio 19096 RTP/AVP 0 96.
a=rtpmap:0 PCMU/8000.
a=rtpmap:96 telephone-event/8000.
a=fmtp:96 0-16.
a=ptime:20.

##################################################################
U 74.67.200.39:33812 -> 222.222.222.222:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 222.222.222.222:5060
;rport=5660;branch=z9hG4bK6pN8Z0pX7y6XD.
From: "ATA" <sip:1276 at domain.company.com>;tag=U4yBUvmKN9Saj.
To: <sip:142 at 172.16.42.13:11852>;tag=1713531477.
Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243932 INVITE.
Contact: <sip:142 at 172.16.42.13:11852>.
Supported: replaces, path, timer, eventlist.
User-Agent: Grandstream Wave/IOS 1.0.19.
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE, MESSAGE.
Content-Type: application/sdp.
Content-Length:   235.
.
v=0.
o=142 8000 8000 IN IP4 172.16.42.13.
s=SIP Call.
c=IN IP4 172.16.42.13.
t=0 0.
m=audio 26390 RTP/AVP 0 8 101.
a=sendrecv.
a=rtpmap:0 PCMU/8000.
a=ptime:20.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.

#
U 222.222.222.222:5060 -> 74.67.200.39:33812
ACK sip:142 at 172.16.42.13:11852 SIP/2.0.
Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bKB4K487aFSBpUQ.
Max-Forwards: 70.
From: "ATA" <sip:1276 at domain.company.com>;tag=U4yBUvmKN9Saj.
To: <sip:142 at 172.16.42.13:11852>;tag=1713531477.
Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243932 ACK.
Contact: <sip:1276 at 222.222.222.222:5060>.
Content-Length: 0.
.

#
U 222.222.222.222:5060 -> 68.5.194.163:12113
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.1.168:12113
;branch=z9hG4bKu22f2-l5MoOcq0;received=68.5.194.163.
From: ATAPHONE <sip:1276 at domain.company.com>;tag=VxLf2-dmKe50.
To: <sip:142 at domain.company.com>;tag=tU5jS13Fr03Qp.
Call-ID: d44p80-lYfd5f2 at domain.company.com.
CSeq: 177 INVITE.
Contact: <sip:142 at 222.222.222.222:5060;transport=udp>.
User-Agent: VOIPGATEWAY.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: path, replaces.
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 219.
Remote-Party-ID: "Outbound Call" <sip:142 at domain.company.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1455140898 1455140899 IN IP4 222.222.222.222.
s=FreeSWITCH.
c=IN IP4 222.222.222.222.
t=0 0.
m=audio 19096 RTP/AVP 0 96.
a=rtpmap:0 PCMU/8000.
a=rtpmap:96 telephone-event/8000.
a=fmtp:96 0-16.
a=ptime:20.

#
U 68.5.194.163:12113 -> 222.222.222.222:5060
ACK sip:142 at 222.222.222.222:5060 SIP/2.0.
From: ATAPHONE<sip:1276 at domain.company.com>;tag=VxLf2-dmKe50.
To: sip:142 at domain.company.com;tag=tU5jS13Fr03Qp.
Call-ID: d44p80-lYfd5f2 at domain.company.com.
CSeq: 177 ACK.
Via: SIP/2.0/UDP 192.168.1.168:12113;branch=z9hG4bKu22f2-VwM5aez*.
Contact: ATAPHONE<sip:1276 at 192.168.1.168:12113>.
Max-Forwards: 70.
User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>.
Content-Length: 0.
.

###########################################################################################
##############################
##############################################################################################################################################################
U 74.67.200.39:33812 -> 222.222.222.222:5060
BYE sip:1276 at 222.222.222.222:5060 SIP/2.0.
Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK364522226;rport.
From: <sip:142 at 172.16.42.13:11852>;tag=1713531477.
To: "ATA" <sip:1276 at domain.company.com>;tag=U4yBUvmKN9Saj.
Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243933 BYE.
Contact: <sip:142 at 172.16.42.13:11852>.
Max-Forwards: 70.
Supported: replaces, path, timer, eventlist.
User-Agent: Grandstream Wave/IOS 1.0.19.
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE, MESSAGE.
Content-Length: 0.
.

#
U 222.222.222.222:5060 -> 74.67.200.39:33812
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 172.16.42.13:11852
;branch=z9hG4bK364522226;rport=33812;received=74.67.200.39.
From: <sip:142 at 172.16.42.13:11852>;tag=1713531477.
To: "ATA" <sip:1276 at domain.company.com>;tag=U4yBUvmKN9Saj.
Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243933 BYE.
User-Agent: VOIPGATEWAY.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: path, replaces.
Content-Length: 0.
.

#
U 222.222.222.222:5060 -> 68.5.194.163:12113
BYE sip:1276 at 192.168.1.168:12113 SIP/2.0.
Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bK89gKvrKN5DvSQ.
Max-Forwards: 70.
From: <sip:142 at domain.company.com>;tag=tU5jS13Fr03Qp.
To: ATAPHONE <sip:1276 at domain.company.com>;tag=VxLf2-dmKe50.
Call-ID: d44p80-lYfd5f2 at domain.company.com.
CSeq: 87243943 BYE.
User-Agent: VOIPGATEWAY.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: path, replaces.
Reason: Q.850;cause=16;text="NORMAL_CLEARING".
Content-Length: 0.

##
U 68.5.194.163:12113 -> 222.222.222.222:5060
SIP/2.0 200 OK.
From: sip:142 at domain.company.com;tag=tU5jS13Fr03Qp.
To: ATAPHONE<sip:1276 at domain.company.com>;tag=VxLf2-dmKe50.
Call-ID: d44p80-lYfd5f2 at domain.company.com.
CSeq: 87243943 BYE.
Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bK89gKvrKN5DvSQ.
User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>.
Content-Length: 0.





On Wed, Feb 10, 2016 at 5:54 PM, ?talo Rossi <italo at freeswitch.org> wrote:

> You need to look at the sip signaling to see what's going on
>
> On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque <shafeeq.v at gmail.com>
> wrote:
>
>> Hello All
>>
>> We are getting one way audio issues with some softphones and grandstream
>> phones behind nat registerd to our freeswitch server.
>>
>> Here is scenario:
>> Grandstream call any extensions (one way audio)
>> Any extension call Grandstream ( Audio works just fine)
>>
>> We have tried multiple softphones and the result is same.
>>
>> Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF
>> issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started
>> with an upgrade to freeswitch.
>>
>> Any help or hint will be much appreciated.
>>
>> Thank you,
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> ?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 jprangi at gmail.com  Thu Feb 11 08:22:55 2016
From: jprangi at gmail.com (Jai Rangi)
Date: Wed, 10 Feb 2016 21:22:55 -0800
Subject: [Freeswitch-users] Mod_callcenter ODBC vs File
In-Reply-To: <CAEJMVkDyLo1iegWVP-gBdtL8LDuvzg0+fBUiru51HFKBfjGqSA@mail.gmail.com>
References: <CAPfjz3gqws5xNFYa8mKfh5NtFrVJo_sCHxqu_a5w3h2a5jcPZw@mail.gmail.com>
	<CAEJMVkDyLo1iegWVP-gBdtL8LDuvzg0+fBUiru51HFKBfjGqSA@mail.gmail.com>
Message-ID: <CAPfjz3iovyUFv18Z3YK9Kpm0+Oc9HCGSqhQ-NV8ifc8r1rm66g@mail.gmail.com>

FYI,
Tested with 1.6.2, 1.6.5. Still the same issue. New call go to abandoned
state immediately.

-Jai


On Wed, Jul 22, 2015 at 6:48 PM, Brian West <brian at freeswitch.org> wrote:

> Once you test our latest release (1.4.20):
>
> [image: Inline image 1]
>
> https://freeswitch.org/jira
>
> Thanks,
>
>
> On Wed, Jul 22, 2015 at 6:58 PM, Jai Rangi <jprangi at gmail.com> wrote:
>
>> Hello,
>>
>> FS version:  FreeSWITCH Version
>> 1.4.18+git~20150312T185523Z~4eed221b69~64bit (git 4eed221 2015-03-12
>> 18:55:23Z 64bit)
>>
>> I found a very strange behavior on mod_call center when used with odbc.
>> This was easy to reproduce when the agents are busy and new calls come in.
>> New member goes in abandoned state immediately and stay in same state
>> forever. Even if previous call is complete and Agent is in Waiting state.
>> Infact some times new members go in abandoned state immediately even if the
>> agents are available.
>>
>> Everything works perfect once I changed the mod_callcenter.conf.xml
>> config to file only mode, new calls go in Waiting state and call is
>> connected to agent as soon as the agent is free.
>>
>> Any idea why the new member goes in abandoned state randomly in place of
>> being in waiting state? Can it be due to too small timeout in call center
>> module as it query the DB every time? Can that be changed?
>>
>> Anyone else had smiler problem, any experience or any recommendations
>> will be much appreciated.
>>
>> Thank you,
>> -Jai
>>
>>
>>
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> 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 krice at freeswitch.org  Thu Feb 11 09:57:13 2016
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 11 Feb 2016 00:57:13 -0600
Subject: [Freeswitch-users] Mod_callcenter ODBC vs File
In-Reply-To: <CAPfjz3iovyUFv18Z3YK9Kpm0+Oc9HCGSqhQ-NV8ifc8r1rm66g@mail.gmail.com>
References: <CAPfjz3gqws5xNFYa8mKfh5NtFrVJo_sCHxqu_a5w3h2a5jcPZw@mail.gmail.com>
	<CAEJMVkDyLo1iegWVP-gBdtL8LDuvzg0+fBUiru51HFKBfjGqSA@mail.gmail.com>
	<CAPfjz3iovyUFv18Z3YK9Kpm0+Oc9HCGSqhQ-NV8ifc8r1rm66g@mail.gmail.com>
Message-ID: <CAPcNkdde+T9ZZR7fNg5BsK3k=1HxbEWc4pKfCR3NzgZjDj_EZA@mail.gmail.com>

is there a jira opened on this? if not, then a dev most likely wont see this


On Wed, Feb 10, 2016 at 11:22 PM, Jai Rangi <jprangi at gmail.com> wrote:

> FYI,
> Tested with 1.6.2, 1.6.5. Still the same issue. New call go to abandoned
> state immediately.
>
> -Jai
>
>
> On Wed, Jul 22, 2015 at 6:48 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Once you test our latest release (1.4.20):
>>
>> [image: Inline image 1]
>>
>> https://freeswitch.org/jira
>>
>> Thanks,
>>
>>
>> On Wed, Jul 22, 2015 at 6:58 PM, Jai Rangi <jprangi at gmail.com> wrote:
>>
>>> Hello,
>>>
>>> FS version:  FreeSWITCH Version
>>> 1.4.18+git~20150312T185523Z~4eed221b69~64bit (git 4eed221 2015-03-12
>>> 18:55:23Z 64bit)
>>>
>>> I found a very strange behavior on mod_call center when used with odbc.
>>> This was easy to reproduce when the agents are busy and new calls come in.
>>> New member goes in abandoned state immediately and stay in same state
>>> forever. Even if previous call is complete and Agent is in Waiting state.
>>> Infact some times new members go in abandoned state immediately even if the
>>> agents are available.
>>>
>>> Everything works perfect once I changed the mod_callcenter.conf.xml
>>> config to file only mode, new calls go in Waiting state and call is
>>> connected to agent as soon as the agent is free.
>>>
>>> Any idea why the new member goes in abandoned state randomly in place of
>>> being in waiting state? Can it be due to too small timeout in call center
>>> module as it query the DB every time? Can that be changed?
>>>
>>> Anyone else had smiler problem, any experience or any recommendations
>>> will be much appreciated.
>>>
>>> Thank you,
>>> -Jai
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>> 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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 jprangi at didforsale.com  Thu Feb 11 10:07:20 2016
From: jprangi at didforsale.com (Jai Rangi)
Date: Wed, 10 Feb 2016 23:07:20 -0800
Subject: [Freeswitch-users] Mod_callcenter ODBC vs File
In-Reply-To: <CAPcNkdde+T9ZZR7fNg5BsK3k=1HxbEWc4pKfCR3NzgZjDj_EZA@mail.gmail.com>
References: <CAPfjz3gqws5xNFYa8mKfh5NtFrVJo_sCHxqu_a5w3h2a5jcPZw@mail.gmail.com>
	<CAEJMVkDyLo1iegWVP-gBdtL8LDuvzg0+fBUiru51HFKBfjGqSA@mail.gmail.com>
	<CAPfjz3iovyUFv18Z3YK9Kpm0+Oc9HCGSqhQ-NV8ifc8r1rm66g@mail.gmail.com>
	<CAPcNkdde+T9ZZR7fNg5BsK3k=1HxbEWc4pKfCR3NzgZjDj_EZA@mail.gmail.com>
Message-ID: <CAPfjz3hJVUYGpdt8iwmxBrVEp69Pg6btdcan+Mb5C6mAhR_jhg@mail.gmail.com>

I can open Jira for that. It might take a day or two. Right now dealing
with another NAT/RTP issue we encountered with the upgrade.
Thank you for your response.
-Jai

*Jai Rangi*
Cebod Technologies LLC dba DIDforSale/Cebod Telecom
O 949-471-0102 <O%20949-471-0101> | C 949-419-7634 <C%20949-742-2666> | F
949-269-0449 / 949-232-1410 | jprangi at didforsale.com  www.cebod.com |
www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA 92626  |






On Wed, Feb 10, 2016 at 10:57 PM, Ken Rice <krice at freeswitch.org> wrote:

> is there a jira opened on this? if not, then a dev most likely wont see
> this
>
>
> On Wed, Feb 10, 2016 at 11:22 PM, Jai Rangi <jprangi at gmail.com> wrote:
>
>> FYI,
>> Tested with 1.6.2, 1.6.5. Still the same issue. New call go to abandoned
>> state immediately.
>>
>> -Jai
>>
>>
>> On Wed, Jul 22, 2015 at 6:48 PM, Brian West <brian at freeswitch.org> wrote:
>>
>>> Once you test our latest release (1.4.20):
>>>
>>> [image: Inline image 1]
>>>
>>> https://freeswitch.org/jira
>>>
>>> Thanks,
>>>
>>>
>>> On Wed, Jul 22, 2015 at 6:58 PM, Jai Rangi <jprangi at gmail.com> wrote:
>>>
>>>> Hello,
>>>>
>>>> FS version:  FreeSWITCH Version
>>>> 1.4.18+git~20150312T185523Z~4eed221b69~64bit (git 4eed221 2015-03-12
>>>> 18:55:23Z 64bit)
>>>>
>>>> I found a very strange behavior on mod_call center when used with odbc.
>>>> This was easy to reproduce when the agents are busy and new calls come in.
>>>> New member goes in abandoned state immediately and stay in same state
>>>> forever. Even if previous call is complete and Agent is in Waiting state.
>>>> Infact some times new members go in abandoned state immediately even if the
>>>> agents are available.
>>>>
>>>> Everything works perfect once I changed the mod_callcenter.conf.xml
>>>> config to file only mode, new calls go in Waiting state and call is
>>>> connected to agent as soon as the agent is free.
>>>>
>>>> Any idea why the new member goes in abandoned state randomly in place
>>>> of being in waiting state? Can it be due to too small timeout in call
>>>> center module as it query the DB every time? Can that be changed?
>>>>
>>>> Anyone else had smiler problem, any experience or any recommendations
>>>> will be much appreciated.
>>>>
>>>> Thank you,
>>>> -Jai
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>>
>>> 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
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 bilaln018 at gmail.com  Thu Feb 11 12:19:16 2016
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Thu, 11 Feb 2016 14:19:16 +0500
Subject: [Freeswitch-users] [SDP Negotiation Error][Calling through JsSIP]
Message-ID: <CADSVerCN05+5vvwvXKY=SrPhPwg8TKAWMUmXBv3P_y=tqmEr6A@mail.gmail.com>

Hi all,

Currently i am facing an CODEC NEGOTIATION ERROR while calling through
JsSIP Caller 1001 Callie 1000,
Please view the logs,
https://pastebin.freeswitch.org/24550

I have enabled PCMA and PCMU in my var.xml,

Regards
Abbasi
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From davidwaf at gmail.com  Thu Feb 11 12:46:06 2016
From: davidwaf at gmail.com (David Wafula)
Date: Thu, 11 Feb 2016 11:46:06 +0200
Subject: [Freeswitch-users] Two Users Registered ..But calls only going one
	way
Message-ID: <CAK9Ws2dOCsLXVS=JPzj4yW-RV_SffNYrWYpoMn=AQy8x_x5gKw@mail.gmail.com>

Hi all,
I have two users who registered in the same domain: user A and B.
A can call B just fine. When B tries to call A, there is silence (no
ringback)..then after sometime the call goes into voice mail. A never
receives the call.  Please see the call trace here:


http://pastebin.com/gWrrS4zw

Am not sure what is causing it.

Regards

-- 
David W
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From steveayre at gmail.com  Thu Feb 11 13:18:18 2016
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 11 Feb 2016 10:18:18 +0000
Subject: [Freeswitch-users] P-asserted-identity
In-Reply-To: <CAArQ-M=aa0rN-pXKe0OXfYy8bQs9qoNVS51TPopu37J0_e2XkA@mail.gmail.com>
References: <CAArQ-M=aa0rN-pXKe0OXfYy8bQs9qoNVS51TPopu37J0_e2XkA@mail.gmail.com>
Message-ID: <CAFiqYu=gYbYsqkvySTJ2qhRXFjWvVFS1r+-zDe8om7S5D8yzGg@mail.gmail.com>

>
> The only thing that works is to set up the "sip_cid_type" variable in the
> dialplan like this:

<action application="bridge" data="{sip_cid_type=pid}sofia/
> gateway/Mygateway/$1"/>


Try setting sip_cid_type=pid as a variable within the gateway definition,
so it's automatically set on every outbound call through that gateway.

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

On 10 February 2016 at 10:31, Jose Serrano <jjserranor at gmail.com> wrote:

> Hello.
>
> My freeswitch  by default replace the P-asserted-identity  by
> Remote-Party-ID when routing calls to the gateway.
> I want to send the P-asserted-identity and not the Remote-Party-ID and for
> that I tried the following:
>
> I have configured  in the outbound gateway definition the following
> parameters:
> <param name="caller-id-type" value="pid"/> or
> <param name="cid-type" value="pid"/>
>  but te behavior is the same.
>
> The only thing that works is to set up the "sip_cid_type" variable in the
> dialplan like this:
> <action application="bridge"
> data="{sip_cid_type=pid}sofia/gateway/Mygateway/$1"/>
>
> Anyone  knows how I can send the P-Asserted-identity without having to
> modify all my dial plan adding the sip_cid_type?
>
> Thanks in avanced
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Thu Feb 11 15:16:02 2016
From: deforceczt at gmail.com (Vladislav Ivanov)
Date: Thu, 11 Feb 2016 14:16:02 +0200
Subject: [Freeswitch-users] Does freeswitch forks his processes?
In-Reply-To: <56B0C9B7.8000106@mst.edu>
References: <CAMCT=3R8CLCX2YBbNghAbr6NouK08=MpXUB2T0puoZ-rhJz-uQ@mail.gmail.com>
	<005d01d15db9$744a3bd0$5cdeb370$@botecomm.com>
	<56B0C6FE.9070507@telefaks.de> <56B0C9B7.8000106@mst.edu>
Message-ID: <CAMCT=3QXHk9p1pHqywozuEiNA2zV-a8b9sYeuCLrpHvjgx05ag@mail.gmail.com>

Hey guys,

It is true, I haven't ran it as root, I fixed it.
But I still have issues with passing more than 50cps, same is happening...
4 cores CPU - CPU is good, 80% idle, load average: 0.81, 0.98
This at 25 cps (I bridge the call, so cps*2)
Then I add 5 cps more:
load average: 1061.40, 333.52
And all OS start's lagging as hell, and i'm unable to find issue, have no
idea what is happening...

My settings:
http://pastebin.com/62B45z4i

Anything I'm missing? Really strange that CPU drops like that...



2016-02-02 17:22 GMT+02:00 Nathan Neulinger <nneul at mst.edu>:

> "Run as" != "Start as"
>
> If you insist on not starting FS as root to let it change user, like most
> other daemons/services, you'll have to jump
> through a bunch of extra steps using file system capabilities to give it
> the ability to set scheduler parameters/etc
> that are restricted to root normally.
>
> -- Nathan
>
> On 02/02/2016 09:10 AM, Peter Steinbach wrote:
> > I've just stumpled over this:
> >  >Is FreeSWITCH starting with root permissions? It needs this in order
> to use the FIFO scheduler and access realtime
> > threads. If not started as root, this would explain your CPS limitations.
> >
> > We like to run Freeswitch as a non privileged user, due to security
> concerns. So there are drawbacks here compared to
> > running FS as root? Can we somehow quantify the differences?
> >
> > Best regards
> > Peter
> >
> >
> > On 02/02/16 13:58, Bote Man wrote:
> >>
> >> Is FreeSWITCH starting with root permissions? It needs this in order to
> use the FIFO scheduler and access realtime
> >> threads. If not started as root, this would explain your CPS
> limitations. There are also limits that can be set in the
> >> config files.
> >>
> >> After it starts it drops privileges to those specified on the command
> line with ?u and ?g switches.
> >>
> >> FreeSWITCH uses multi-threading. I do not know about htop, but maybe it
> is showing the multiple threads?
> >>
> >> top ?H shows each thread.
> >>
> >> ---
> >>
> >> Bote
> >>
> >> FreeSWITCH Docs Janitor
> >>
> >> http://freeswitch.org/confluence
> >>
> >> *From:*Vladislav Ivanov
> >> *Sent:* Tuesday, 02 February, 2016 07:09
> >> *Subject:* [Freeswitch-users] Does freeswitch forks his processes?
> >>
> >> Hey guys,
> >>
> >> I have a question about freeswitch process/threading usage.
> >> So far that I haven't noticed freeswitch to fork himself, I have only 1
> freeswitch instance.
> >> http://i.imgur.com/bdbYOwp.png
> >>
> >> But then I found screenshot of htop with freeswitch and noticed that
> there is multiple freeswitch processes being run:
> >> http://i.imgur.com/VNpl55z.jpg
> >>
> >> I'm having issues with "loading" the freeswitch after 50 cps in any
> cpu/ram configuration.
> >> Be it physical or virtual environment I cant pass the 50 cps mark.
> >> I have strange issue with CPU usage on same CPS:
> >>
> >> http://i.imgur.com/8BdQWVL.png
> >> http://i.imgur.com/mWRnoGr.png
> >>
> >> I timeload test freeswitch with 50cps for 5+ hours, and seems like
> there is some kind of leak somewhere.
> >> I have tested configuration on:
> >> Debian 8
> >> 2 core/8 gb ram
> >> 4 core/8 gb ram (graphs are from here)
> >> 8 core/32 gb ram
> >>
> >> and in all the tests I were not able to send more than 50 cps without
> CPU dropping to 0 with all system starting to
> >> respond really laggy.
> >>
> >> Test is:
> >> sipp -> freeswitch -> sipp
> >>
> >> Just 1 dialpeer with bridge action. No gateways. Just simple dialplan
> and 1 profile...
> >> Any advice?
> >>
> >> Thank you all
> >>
> >>
> >>
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > --
> > With kind regards
> > Peter Steinbach
> >
> > Telefaks Services GmbH
> > mailto:lists  (att) telefaks.de
> > Internet:www.telefaks.de
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> --
> ------------------------------------------------------------
> Nathan Neulinger                       nneul at mst.edu
> Missouri S&T Information Technology    (573) 612-1412
> System Administrator - Architect
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gmaruzz at gmail.com  Thu Feb 11 16:57:10 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 11 Feb 2016 14:57:10 +0100
Subject: [Freeswitch-users] Does freeswitch forks his processes?
In-Reply-To: <CAMCT=3QXHk9p1pHqywozuEiNA2zV-a8b9sYeuCLrpHvjgx05ag@mail.gmail.com>
References: <CAMCT=3R8CLCX2YBbNghAbr6NouK08=MpXUB2T0puoZ-rhJz-uQ@mail.gmail.com>
	<005d01d15db9$744a3bd0$5cdeb370$@botecomm.com>
	<56B0C6FE.9070507@telefaks.de> <56B0C9B7.8000106@mst.edu>
	<CAMCT=3QXHk9p1pHqywozuEiNA2zV-a8b9sYeuCLrpHvjgx05ag@mail.gmail.com>
Message-ID: <CALXCt0ro+z3hyKCStZB36CDjNt=n_Dk2Vn039Y1gZUCaqD8sGQ@mail.gmail.com>

by default FreeSWITCH has a limit of 30cps... You must edit config files to
change that...



On Thu, Feb 11, 2016 at 1:16 PM, Vladislav Ivanov <deforceczt at gmail.com>
wrote:

> Hey guys,
>
> It is true, I haven't ran it as root, I fixed it.
> But I still have issues with passing more than 50cps, same is happening...
> 4 cores CPU - CPU is good, 80% idle, load average: 0.81, 0.98
> This at 25 cps (I bridge the call, so cps*2)
> Then I add 5 cps more:
> load average: 1061.40, 333.52
> And all OS start's lagging as hell, and i'm unable to find issue, have no
> idea what is happening...
>
> My settings:
> http://pastebin.com/62B45z4i
>
> Anything I'm missing? Really strange that CPU drops like that...
>
>
>
> 2016-02-02 17:22 GMT+02:00 Nathan Neulinger <nneul at mst.edu>:
>
>> "Run as" != "Start as"
>>
>> If you insist on not starting FS as root to let it change user, like most
>> other daemons/services, you'll have to jump
>> through a bunch of extra steps using file system capabilities to give it
>> the ability to set scheduler parameters/etc
>> that are restricted to root normally.
>>
>> -- Nathan
>>
>> On 02/02/2016 09:10 AM, Peter Steinbach wrote:
>> > I've just stumpled over this:
>> >  >Is FreeSWITCH starting with root permissions? It needs this in order
>> to use the FIFO scheduler and access realtime
>> > threads. If not started as root, this would explain your CPS
>> limitations.
>> >
>> > We like to run Freeswitch as a non privileged user, due to security
>> concerns. So there are drawbacks here compared to
>> > running FS as root? Can we somehow quantify the differences?
>> >
>> > Best regards
>> > Peter
>> >
>> >
>> > On 02/02/16 13:58, Bote Man wrote:
>> >>
>> >> Is FreeSWITCH starting with root permissions? It needs this in order
>> to use the FIFO scheduler and access realtime
>> >> threads. If not started as root, this would explain your CPS
>> limitations. There are also limits that can be set in the
>> >> config files.
>> >>
>> >> After it starts it drops privileges to those specified on the command
>> line with ?u and ?g switches.
>> >>
>> >> FreeSWITCH uses multi-threading. I do not know about htop, but maybe
>> it is showing the multiple threads?
>> >>
>> >> top ?H shows each thread.
>> >>
>> >> ---
>> >>
>> >> Bote
>> >>
>> >> FreeSWITCH Docs Janitor
>> >>
>> >> http://freeswitch.org/confluence
>> >>
>> >> *From:*Vladislav Ivanov
>> >> *Sent:* Tuesday, 02 February, 2016 07:09
>> >> *Subject:* [Freeswitch-users] Does freeswitch forks his processes?
>> >>
>> >> Hey guys,
>> >>
>> >> I have a question about freeswitch process/threading usage.
>> >> So far that I haven't noticed freeswitch to fork himself, I have only
>> 1 freeswitch instance.
>> >> http://i.imgur.com/bdbYOwp.png
>> >>
>> >> But then I found screenshot of htop with freeswitch and noticed that
>> there is multiple freeswitch processes being run:
>> >> http://i.imgur.com/VNpl55z.jpg
>> >>
>> >> I'm having issues with "loading" the freeswitch after 50 cps in any
>> cpu/ram configuration.
>> >> Be it physical or virtual environment I cant pass the 50 cps mark.
>> >> I have strange issue with CPU usage on same CPS:
>> >>
>> >> http://i.imgur.com/8BdQWVL.png
>> >> http://i.imgur.com/mWRnoGr.png
>> >>
>> >> I timeload test freeswitch with 50cps for 5+ hours, and seems like
>> there is some kind of leak somewhere.
>> >> I have tested configuration on:
>> >> Debian 8
>> >> 2 core/8 gb ram
>> >> 4 core/8 gb ram (graphs are from here)
>> >> 8 core/32 gb ram
>> >>
>> >> and in all the tests I were not able to send more than 50 cps without
>> CPU dropping to 0 with all system starting to
>> >> respond really laggy.
>> >>
>> >> Test is:
>> >> sipp -> freeswitch -> sipp
>> >>
>> >> Just 1 dialpeer with bridge action. No gateways. Just simple dialplan
>> and 1 profile...
>> >> Any advice?
>> >>
>> >> Thank you all
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://confluence.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > --
>> > With kind regards
>> > Peter Steinbach
>> >
>> > Telefaks Services GmbH
>> > mailto:lists  (att) telefaks.de
>> > Internet:www.telefaks.de
>> >
>> >
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> --
>> ------------------------------------------------------------
>> Nathan Neulinger                       nneul at mst.edu
>> Missouri S&T Information Technology    (573) 612-1412
>> System Administrator - Architect
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 luis.daniel.lucio at gmail.com  Thu Feb 11 05:14:39 2016
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Wed, 10 Feb 2016 21:14:39 -0500
Subject: [Freeswitch-users] Oneway audio issues in freeswitch
In-Reply-To: <CABMX_w_NWvQ5ZUdxtYHnua4FD9i4hYDt3YeP-X+43XYi4cb=8A@mail.gmail.com>
References: <CANw+5nWmd_j=eUz7OGoH=5Wb3g43VOS_XF_dZVkuFRMRUtfZ2Q@mail.gmail.com>
	<CABMX_w_NWvQ5ZUdxtYHnua4FD9i4hYDt3YeP-X+43XYi4cb=8A@mail.gmail.com>
Message-ID: <CAFLo2QzdR1YBLv6-y5hs=juyE5N2uZw+N5dCiyq4S3Hbgkb2eg@mail.gmail.com>

As a rule of dumb, try turning on rport
Le 10 f?vr. 2016 8:55 PM, "?talo Rossi" <italo at freeswitch.org> a ?crit :

> You need to look at the sip signaling to see what's going on
>
> On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque <shafeeq.v at gmail.com>
> wrote:
>
>> Hello All
>>
>> We are getting one way audio issues with some softphones and grandstream
>> phones behind nat registerd to our freeswitch server.
>>
>> Here is scenario:
>> Grandstream call any extensions (one way audio)
>> Any extension call Grandstream ( Audio works just fine)
>>
>> We have tried multiple softphones and the result is same.
>>
>> Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF
>> issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started
>> with an upgrade to freeswitch.
>>
>> Any help or hint will be much appreciated.
>>
>> Thank you,
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> ?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 mailings at interloop-software.de  Thu Feb 11 16:54:02 2016
From: mailings at interloop-software.de (Dominik Steinbrecher)
Date: Thu, 11 Feb 2016 14:54:02 +0100
Subject: [Freeswitch-users] Jessie: Unable to locate package
Message-ID: <CCB34BD8-C5CA-47DF-B220-48079FEECEF3@interloop-software.de>

Hi, 

I want to install freeswitch 1.6 on a freshly installed Debian 8.3 jessie. 

I followed the guide from:
https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie

When I execute apt-get install I got the following errors:

root at freeswitch:~# apt-get update && apt-get install -y freeswitch-all freeswitch-all-dbg gdb
Hit http://security.debian.org jessie/updates InRelease
Ign http://ftp.de.debian.org jessie InRelease                                         
Hit http://ftp.de.debian.org jessie-updates InRelease                                                  
Hit http://ftp.de.debian.org jessie Release.gpg                                                        
Hit http://security.debian.org jessie/updates/main Sources                                                                  
Hit http://ftp.de.debian.org jessie Release                                                                                 
Hit http://security.debian.org jessie/updates/main i386 Packages                                                                           
Hit http://security.debian.org jessie/updates/main Translation-en                                                                          
Hit http://ftp.de.debian.org jessie-updates/main Sources                              
Get:1 http://ftp.de.debian.org jessie-updates/main i386 Packages/DiffIndex [367 B]
Hit http://ftp.de.debian.org jessie-updates/main Translation-en     
Hit http://ftp.de.debian.org jessie/main Sources                    
Hit http://ftp.de.debian.org jessie/main i386 Packages               
Hit http://ftp.de.debian.org jessie/main Translation-en                 
Hit http://files.freeswitch.org jessie InRelease                        
Hit http://files.freeswitch.org jessie/main i386 Packages
Ign http://files.freeswitch.org jessie/main Translation-en_US
Ign http://files.freeswitch.org jessie/main Translation-en
Fetched 367 B in 3s (102 B/s)
Reading package lists... Done
Reading package lists... Done
Building dependency tree       
Reading state information... Done
E: Unable to locate package freeswitch-all
E: Unable to locate package freeswitch-all-dbg

I searched for the freeswitch-all package with aptitude but there?s no such package.

Any ideas, what I am doing wrong? Is there anything I could try or check?

The debian jessie 8.3 64bit is running as VM under VirtualBox. The host computer is a Mac Mini with OS X 10.11.1. It?s a freshly installed VM just for testing freeswitch. 

Thanks a lot
Dominik



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From gmaruzz at gmail.com  Thu Feb 11 17:20:20 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 11 Feb 2016 15:20:20 +0100
Subject: [Freeswitch-users] Jessie: Unable to locate package
In-Reply-To: <CCB34BD8-C5CA-47DF-B220-48079FEECEF3@interloop-software.de>
References: <CCB34BD8-C5CA-47DF-B220-48079FEECEF3@interloop-software.de>
Message-ID: <CALXCt0rDMs1DVO8Hdh=bcbX89ugmyPTqWej3srKqEZhGhg+PFA@mail.gmail.com>

are you using an i386 VM?

You better use a 64bit VM... (not sure 32 bit is supported)


On Thu, Feb 11, 2016 at 2:54 PM, Dominik Steinbrecher <
mailings at interloop-software.de> wrote:

> Hi,
>
> I want to install freeswitch 1.6 on a freshly installed Debian 8.3 jessie.
>
> I followed the guide from:
> https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie
>
> When I execute apt-get install I got the following errors:
>
> root at freeswitch:~# apt-get update && apt-get install -y freeswitch-all
> freeswitch-all-dbg gdb
> Hit http://security.debian.org jessie/updates InRelease
> Ign http://ftp.de.debian.org jessie InRelease
>
> Hit http://ftp.de.debian.org jessie-updates InRelease
>
> Hit http://ftp.de.debian.org jessie Release.gpg
>
> Hit http://security.debian.org jessie/updates/main Sources
>
> Hit http://ftp.de.debian.org jessie Release
>
> Hit http://security.debian.org jessie/updates/main i386 Packages
>
> Hit http://security.debian.org jessie/updates/main Translation-en
>
> Hit http://ftp.de.debian.org jessie-updates/main Sources
>
> Get:1 http://ftp.de.debian.org jessie-updates/main i386
> Packages/DiffIndex [367 B]
> Hit http://ftp.de.debian.org jessie-updates/main Translation-en
> Hit http://ftp.de.debian.org jessie/main Sources
> Hit http://ftp.de.debian.org jessie/main i386 Packages
> Hit http://ftp.de.debian.org jessie/main Translation-en
> Hit http://files.freeswitch.org jessie InRelease
> Hit http://files.freeswitch.org jessie/main i386 Packages
> Ign http://files.freeswitch.org jessie/main Translation-en_US
> Ign http://files.freeswitch.org jessie/main Translation-en
> Fetched 367 B in 3s (102 B/s)
> Reading package lists... Done
> Reading package lists... Done
> Building dependency tree
> Reading state information... Done
> E: Unable to locate package freeswitch-all
> E: Unable to locate package freeswitch-all-dbg
>
> I searched for the freeswitch-all package with aptitude but there?s no
> such package.
>
> Any ideas, what I am doing wrong? Is there anything I could try or check?
>
> The debian jessie 8.3 64bit is running as VM under VirtualBox. The host
> computer is a Mac Mini with OS X 10.11.1. It?s a freshly installed VM just
> for testing freeswitch.
>
> Thanks a lot
> Dominik
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 regis.freeswitch.org at tornad.net  Thu Feb 11 17:32:13 2016
From: regis.freeswitch.org at tornad.net (Regis M)
Date: Thu, 11 Feb 2016 15:32:13 +0100
Subject: [Freeswitch-users] Jessie: Unable to locate package
In-Reply-To: <CALXCt0rDMs1DVO8Hdh=bcbX89ugmyPTqWej3srKqEZhGhg+PFA@mail.gmail.com>
References: <CCB34BD8-C5CA-47DF-B220-48079FEECEF3@interloop-software.de>
	<CALXCt0rDMs1DVO8Hdh=bcbX89ugmyPTqWej3srKqEZhGhg+PFA@mail.gmail.com>
Message-ID: <CAHWtr1QGU2fgQJeG8RTfyBv1Ov=cFeQY_XSOxMNm6NBFQiROEQ@mail.gmail.com>

I confirm, I install 4 or 5 of them each day since 3 days for deployement
tests and I have not problem with package 1.6 Debian 8.3 x64

Regard,


2016-02-11 15:20 GMT+01:00 Giovanni Maruzzelli <gmaruzz at gmail.com>:

> are you using an i386 VM?
>
> You better use a 64bit VM... (not sure 32 bit is supported)
>
>
> On Thu, Feb 11, 2016 at 2:54 PM, Dominik Steinbrecher <
> mailings at interloop-software.de> wrote:
>
>> Hi,
>>
>> I want to install freeswitch 1.6 on a freshly installed Debian 8.3
>> jessie.
>>
>> I followed the guide from:
>> https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie
>>
>> When I execute apt-get install I got the following errors:
>>
>> root at freeswitch:~# apt-get update && apt-get install -y freeswitch-all
>> freeswitch-all-dbg gdb
>> Hit http://security.debian.org jessie/updates InRelease
>> Ign http://ftp.de.debian.org jessie InRelease
>>
>> Hit http://ftp.de.debian.org jessie-updates InRelease
>>
>> Hit http://ftp.de.debian.org jessie Release.gpg
>>
>> Hit http://security.debian.org jessie/updates/main Sources
>>
>> Hit http://ftp.de.debian.org jessie Release
>>
>> Hit http://security.debian.org jessie/updates/main i386 Packages
>>
>> Hit http://security.debian.org jessie/updates/main Translation-en
>>
>> Hit http://ftp.de.debian.org jessie-updates/main Sources
>>
>> Get:1 http://ftp.de.debian.org jessie-updates/main i386
>> Packages/DiffIndex [367 B]
>> Hit http://ftp.de.debian.org jessie-updates/main Translation-en
>> Hit http://ftp.de.debian.org jessie/main Sources
>> Hit http://ftp.de.debian.org jessie/main i386 Packages
>> Hit http://ftp.de.debian.org jessie/main Translation-en
>> Hit http://files.freeswitch.org jessie InRelease
>> Hit http://files.freeswitch.org jessie/main i386 Packages
>> Ign http://files.freeswitch.org jessie/main Translation-en_US
>> Ign http://files.freeswitch.org jessie/main Translation-en
>> Fetched 367 B in 3s (102 B/s)
>> Reading package lists... Done
>> Reading package lists... Done
>> Building dependency tree
>> Reading state information... Done
>> E: Unable to locate package freeswitch-all
>> E: Unable to locate package freeswitch-all-dbg
>>
>> I searched for the freeswitch-all package with aptitude but there?s no
>> such package.
>>
>> Any ideas, what I am doing wrong? Is there anything I could try or check?
>>
>> The debian jessie 8.3 64bit is running as VM under VirtualBox. The host
>> computer is a Mac Mini with OS X 10.11.1. It?s a freshly installed VM just
>> for testing freeswitch.
>>
>> Thanks a lot
>> Dominik
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 brian at freeswitch.org  Thu Feb 11 17:37:45 2016
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Feb 2016 08:37:45 -0600
Subject: [Freeswitch-users] [SDP Negotiation Error][Calling through
	JsSIP]
In-Reply-To: <CADSVerCN05+5vvwvXKY=SrPhPwg8TKAWMUmXBv3P_y=tqmEr6A@mail.gmail.com>
References: <CADSVerCN05+5vvwvXKY=SrPhPwg8TKAWMUmXBv3P_y=tqmEr6A@mail.gmail.com>
Message-ID: <CAEJMVkCrUZwSUhjM6s4+H5N28=i_0Qj+FoqjuAiDzYW0D1FCUA@mail.gmail.com>

NO candidate ACL defined, Defaulting to wan.auto
sofia/internal/1001 at 192.241.213.201:7000 no suitable candidates found.

On Thu, Feb 11, 2016 at 3:19 AM, Bilal Abbasi <bilaln018 at gmail.com> wrote:

> Hi all,
>
> Currently i am facing an CODEC NEGOTIATION ERROR while calling through
> JsSIP Caller 1001 Callie 1000,
> Please view the logs,
> https://pastebin.freeswitch.org/24550
>
> I have enabled PCMA and PCMU in my var.xml,
>
> Regards
> Abbasi
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


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

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  Thu Feb 11 17:38:39 2016
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Feb 2016 08:38:39 -0600
Subject: [Freeswitch-users] Does freeswitch forks his processes?
In-Reply-To: <CAMCT=3QXHk9p1pHqywozuEiNA2zV-a8b9sYeuCLrpHvjgx05ag@mail.gmail.com>
References: <CAMCT=3R8CLCX2YBbNghAbr6NouK08=MpXUB2T0puoZ-rhJz-uQ@mail.gmail.com>
	<005d01d15db9$744a3bd0$5cdeb370$@botecomm.com>
	<56B0C6FE.9070507@telefaks.de> <56B0C9B7.8000106@mst.edu>
	<CAMCT=3QXHk9p1pHqywozuEiNA2zV-a8b9sYeuCLrpHvjgx05ag@mail.gmail.com>
Message-ID: <CAEJMVkA6YoUwOmmsNy6P6rNVebczY=K6ww2-GnBuDopc+a_=UQ@mail.gmail.com>

I sense that your load testing method is flawed and you're just DDoSing the
box.

On Thu, Feb 11, 2016 at 6:16 AM, Vladislav Ivanov <deforceczt at gmail.com>
wrote:

> Hey guys,
>
> It is true, I haven't ran it as root, I fixed it.
> But I still have issues with passing more than 50cps, same is happening...
> 4 cores CPU - CPU is good, 80% idle, load average: 0.81, 0.98
> This at 25 cps (I bridge the call, so cps*2)
> Then I add 5 cps more:
> load average: 1061.40, 333.52
> And all OS start's lagging as hell, and i'm unable to find issue, have no
> idea what is happening...
>
> My settings:
> http://pastebin.com/62B45z4i
>
> Anything I'm missing? Really strange that CPU drops like that...
>
>
>
> 2016-02-02 17:22 GMT+02:00 Nathan Neulinger <nneul at mst.edu>:
>
>> "Run as" != "Start as"
>>
>> If you insist on not starting FS as root to let it change user, like most
>> other daemons/services, you'll have to jump
>> through a bunch of extra steps using file system capabilities to give it
>> the ability to set scheduler parameters/etc
>> that are restricted to root normally.
>>
>> -- Nathan
>>
>> On 02/02/2016 09:10 AM, Peter Steinbach wrote:
>> > I've just stumpled over this:
>> >  >Is FreeSWITCH starting with root permissions? It needs this in order
>> to use the FIFO scheduler and access realtime
>> > threads. If not started as root, this would explain your CPS
>> limitations.
>> >
>> > We like to run Freeswitch as a non privileged user, due to security
>> concerns. So there are drawbacks here compared to
>> > running FS as root? Can we somehow quantify the differences?
>> >
>> > Best regards
>> > Peter
>> >
>> >
>> > On 02/02/16 13:58, Bote Man wrote:
>> >>
>> >> Is FreeSWITCH starting with root permissions? It needs this in order
>> to use the FIFO scheduler and access realtime
>> >> threads. If not started as root, this would explain your CPS
>> limitations. There are also limits that can be set in the
>> >> config files.
>> >>
>> >> After it starts it drops privileges to those specified on the command
>> line with ?u and ?g switches.
>> >>
>> >> FreeSWITCH uses multi-threading. I do not know about htop, but maybe
>> it is showing the multiple threads?
>> >>
>> >> top ?H shows each thread.
>> >>
>> >> ---
>> >>
>> >> Bote
>> >>
>> >> FreeSWITCH Docs Janitor
>> >>
>> >> http://freeswitch.org/confluence
>> >>
>> >> *From:*Vladislav Ivanov
>> >> *Sent:* Tuesday, 02 February, 2016 07:09
>> >> *Subject:* [Freeswitch-users] Does freeswitch forks his processes?
>> >>
>> >> Hey guys,
>> >>
>> >> I have a question about freeswitch process/threading usage.
>> >> So far that I haven't noticed freeswitch to fork himself, I have only
>> 1 freeswitch instance.
>> >> http://i.imgur.com/bdbYOwp.png
>> >>
>> >> But then I found screenshot of htop with freeswitch and noticed that
>> there is multiple freeswitch processes being run:
>> >> http://i.imgur.com/VNpl55z.jpg
>> >>
>> >> I'm having issues with "loading" the freeswitch after 50 cps in any
>> cpu/ram configuration.
>> >> Be it physical or virtual environment I cant pass the 50 cps mark.
>> >> I have strange issue with CPU usage on same CPS:
>> >>
>> >> http://i.imgur.com/8BdQWVL.png
>> >> http://i.imgur.com/mWRnoGr.png
>> >>
>> >> I timeload test freeswitch with 50cps for 5+ hours, and seems like
>> there is some kind of leak somewhere.
>> >> I have tested configuration on:
>> >> Debian 8
>> >> 2 core/8 gb ram
>> >> 4 core/8 gb ram (graphs are from here)
>> >> 8 core/32 gb ram
>> >>
>> >> and in all the tests I were not able to send more than 50 cps without
>> CPU dropping to 0 with all system starting to
>> >> respond really laggy.
>> >>
>> >> Test is:
>> >> sipp -> freeswitch -> sipp
>> >>
>> >> Just 1 dialpeer with bridge action. No gateways. Just simple dialplan
>> and 1 profile...
>> >> Any advice?
>> >>
>> >> Thank you all
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://confluence.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > --
>> > With kind regards
>> > Peter Steinbach
>> >
>> > Telefaks Services GmbH
>> > mailto:lists  (att) telefaks.de
>> > Internet:www.telefaks.de
>> >
>> >
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> --
>> ------------------------------------------------------------
>> Nathan Neulinger                       nneul at mst.edu
>> Missouri S&T Information Technology    (573) 612-1412
>> System Administrator - Architect
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

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 italo at freeswitch.org  Thu Feb 11 17:38:53 2016
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Thu, 11 Feb 2016 11:38:53 -0300
Subject: [Freeswitch-users] Mod_callcenter ODBC vs File
In-Reply-To: <CAPfjz3hJVUYGpdt8iwmxBrVEp69Pg6btdcan+Mb5C6mAhR_jhg@mail.gmail.com>
References: <CAPfjz3gqws5xNFYa8mKfh5NtFrVJo_sCHxqu_a5w3h2a5jcPZw@mail.gmail.com>
	<CAEJMVkDyLo1iegWVP-gBdtL8LDuvzg0+fBUiru51HFKBfjGqSA@mail.gmail.com>
	<CAPfjz3iovyUFv18Z3YK9Kpm0+Oc9HCGSqhQ-NV8ifc8r1rm66g@mail.gmail.com>
	<CAPcNkdde+T9ZZR7fNg5BsK3k=1HxbEWc4pKfCR3NzgZjDj_EZA@mail.gmail.com>
	<CAPfjz3hJVUYGpdt8iwmxBrVEp69Pg6btdcan+Mb5C6mAhR_jhg@mail.gmail.com>
Message-ID: <CABMX_w8DSg+co8W7=XHk0zf6401s1X7xynjwWqCU+x_3PaGaAA@mail.gmail.com>

Give me a debug log, your odbc settings and callcenter configs so I can
take a quick look

On Thu, Feb 11, 2016 at 4:07 AM, Jai Rangi <jprangi at didforsale.com> wrote:

> I can open Jira for that. It might take a day or two. Right now dealing
> with another NAT/RTP issue we encountered with the upgrade.
> Thank you for your response.
> -Jai
>
> *Jai Rangi*
> Cebod Technologies LLC dba DIDforSale/Cebod Telecom
> O 949-471-0102 <O%20949-471-0101> | C 949-419-7634 <C%20949-742-2666> | F
> 949-269-0449 / 949-232-1410 | jprangi at didforsale.com  www.cebod.com |
> www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA 92626  |
>
>
>
>
>
>
> On Wed, Feb 10, 2016 at 10:57 PM, Ken Rice <krice at freeswitch.org> wrote:
>
>> is there a jira opened on this? if not, then a dev most likely wont see
>> this
>>
>>
>> On Wed, Feb 10, 2016 at 11:22 PM, Jai Rangi <jprangi at gmail.com> wrote:
>>
>>> FYI,
>>> Tested with 1.6.2, 1.6.5. Still the same issue. New call go to abandoned
>>> state immediately.
>>>
>>> -Jai
>>>
>>>
>>> On Wed, Jul 22, 2015 at 6:48 PM, Brian West <brian at freeswitch.org>
>>> wrote:
>>>
>>>> Once you test our latest release (1.4.20):
>>>>
>>>> [image: Inline image 1]
>>>>
>>>> https://freeswitch.org/jira
>>>>
>>>> Thanks,
>>>>
>>>>
>>>> On Wed, Jul 22, 2015 at 6:58 PM, Jai Rangi <jprangi at gmail.com> wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>> FS version:  FreeSWITCH Version
>>>>> 1.4.18+git~20150312T185523Z~4eed221b69~64bit (git 4eed221 2015-03-12
>>>>> 18:55:23Z 64bit)
>>>>>
>>>>> I found a very strange behavior on mod_call center when used with
>>>>> odbc. This was easy to reproduce when the agents are busy and new calls
>>>>> come in. New member goes in abandoned state immediately and stay in same
>>>>> state forever. Even if previous call is complete and Agent is in Waiting
>>>>> state. Infact some times new members go in abandoned state immediately even
>>>>> if the agents are available.
>>>>>
>>>>> Everything works perfect once I changed the mod_callcenter.conf.xml
>>>>> config to file only mode, new calls go in Waiting state and call is
>>>>> connected to agent as soon as the agent is free.
>>>>>
>>>>> Any idea why the new member goes in abandoned state randomly in place
>>>>> of being in waiting state? Can it be due to too small timeout in call
>>>>> center module as it query the DB every time? Can that be changed?
>>>>>
>>>>> Anyone else had smiler problem, any experience or any recommendations
>>>>> will be much appreciated.
>>>>>
>>>>> Thank you,
>>>>> -Jai
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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
>>>>
>>>> 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
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From deforceczt at gmail.com  Thu Feb 11 17:45:24 2016
From: deforceczt at gmail.com (Vladislav Ivanov)
Date: Thu, 11 Feb 2016 16:45:24 +0200
Subject: [Freeswitch-users] Does freeswitch forks his processes?
In-Reply-To: <CAEJMVkA6YoUwOmmsNy6P6rNVebczY=K6ww2-GnBuDopc+a_=UQ@mail.gmail.com>
References: <CAMCT=3R8CLCX2YBbNghAbr6NouK08=MpXUB2T0puoZ-rhJz-uQ@mail.gmail.com>
	<005d01d15db9$744a3bd0$5cdeb370$@botecomm.com>
	<56B0C6FE.9070507@telefaks.de> <56B0C9B7.8000106@mst.edu>
	<CAMCT=3QXHk9p1pHqywozuEiNA2zV-a8b9sYeuCLrpHvjgx05ag@mail.gmail.com>
	<CAEJMVkA6YoUwOmmsNy6P6rNVebczY=K6ww2-GnBuDopc+a_=UQ@mail.gmail.com>
Message-ID: <CAMCT=3Q03rBbL6Gw2Jtkiyw_mNZbkVEtP43j1d8OgVfRHQ5HeA@mail.gmail.com>

I have changed it:
 <param name="sessions-per-second" value="400"/>

If there was limit - there would be no CPU hogging.
Here is status command:
FreeSWITCH (Version 1.6.6 git d2d0b32 2016-01-11 20:16:12Z 64bit) is ready
0 session(s) since startup
0 session(s) - peak 0, last 5min 0
0 session(s) per Sec out of max 400, peak 0, last 5min 0
10000 session(s) max
min idle cpu 0.00/100.00
Current Stack Size/Max 240K/8192K

I can believe that i'm just DDoSing my box.
I'm using simple uac/uas from sipp, and certain load is ok, but after that
it goes straight down with increase of just 5 cps... I mean it is strange
that with 20 cps it's 25% loaded and with 25 cps it's 100% loaded.

2016-02-11 16:38 GMT+02:00 Brian West <brian at freeswitch.org>:

> I sense that your load testing method is flawed and you're just DDoSing
> the box.
>
> On Thu, Feb 11, 2016 at 6:16 AM, Vladislav Ivanov <deforceczt at gmail.com>
> wrote:
>
>> Hey guys,
>>
>> It is true, I haven't ran it as root, I fixed it.
>> But I still have issues with passing more than 50cps, same is
>> happening...
>> 4 cores CPU - CPU is good, 80% idle, load average: 0.81, 0.98
>> This at 25 cps (I bridge the call, so cps*2)
>> Then I add 5 cps more:
>> load average: 1061.40, 333.52
>> And all OS start's lagging as hell, and i'm unable to find issue, have no
>> idea what is happening...
>>
>> My settings:
>> http://pastebin.com/62B45z4i
>>
>> Anything I'm missing? Really strange that CPU drops like that...
>>
>>
>>
>> 2016-02-02 17:22 GMT+02:00 Nathan Neulinger <nneul at mst.edu>:
>>
>>> "Run as" != "Start as"
>>>
>>> If you insist on not starting FS as root to let it change user, like
>>> most other daemons/services, you'll have to jump
>>> through a bunch of extra steps using file system capabilities to give it
>>> the ability to set scheduler parameters/etc
>>> that are restricted to root normally.
>>>
>>> -- Nathan
>>>
>>> On 02/02/2016 09:10 AM, Peter Steinbach wrote:
>>> > I've just stumpled over this:
>>> >  >Is FreeSWITCH starting with root permissions? It needs this in order
>>> to use the FIFO scheduler and access realtime
>>> > threads. If not started as root, this would explain your CPS
>>> limitations.
>>> >
>>> > We like to run Freeswitch as a non privileged user, due to security
>>> concerns. So there are drawbacks here compared to
>>> > running FS as root? Can we somehow quantify the differences?
>>> >
>>> > Best regards
>>> > Peter
>>> >
>>> >
>>> > On 02/02/16 13:58, Bote Man wrote:
>>> >>
>>> >> Is FreeSWITCH starting with root permissions? It needs this in order
>>> to use the FIFO scheduler and access realtime
>>> >> threads. If not started as root, this would explain your CPS
>>> limitations. There are also limits that can be set in the
>>> >> config files.
>>> >>
>>> >> After it starts it drops privileges to those specified on the command
>>> line with ?u and ?g switches.
>>> >>
>>> >> FreeSWITCH uses multi-threading. I do not know about htop, but maybe
>>> it is showing the multiple threads?
>>> >>
>>> >> top ?H shows each thread.
>>> >>
>>> >> ---
>>> >>
>>> >> Bote
>>> >>
>>> >> FreeSWITCH Docs Janitor
>>> >>
>>> >> http://freeswitch.org/confluence
>>> >>
>>> >> *From:*Vladislav Ivanov
>>> >> *Sent:* Tuesday, 02 February, 2016 07:09
>>> >> *Subject:* [Freeswitch-users] Does freeswitch forks his processes?
>>> >>
>>> >> Hey guys,
>>> >>
>>> >> I have a question about freeswitch process/threading usage.
>>> >> So far that I haven't noticed freeswitch to fork himself, I have only
>>> 1 freeswitch instance.
>>> >> http://i.imgur.com/bdbYOwp.png
>>> >>
>>> >> But then I found screenshot of htop with freeswitch and noticed that
>>> there is multiple freeswitch processes being run:
>>> >> http://i.imgur.com/VNpl55z.jpg
>>> >>
>>> >> I'm having issues with "loading" the freeswitch after 50 cps in any
>>> cpu/ram configuration.
>>> >> Be it physical or virtual environment I cant pass the 50 cps mark.
>>> >> I have strange issue with CPU usage on same CPS:
>>> >>
>>> >> http://i.imgur.com/8BdQWVL.png
>>> >> http://i.imgur.com/mWRnoGr.png
>>> >>
>>> >> I timeload test freeswitch with 50cps for 5+ hours, and seems like
>>> there is some kind of leak somewhere.
>>> >> I have tested configuration on:
>>> >> Debian 8
>>> >> 2 core/8 gb ram
>>> >> 4 core/8 gb ram (graphs are from here)
>>> >> 8 core/32 gb ram
>>> >>
>>> >> and in all the tests I were not able to send more than 50 cps without
>>> CPU dropping to 0 with all system starting to
>>> >> respond really laggy.
>>> >>
>>> >> Test is:
>>> >> sipp -> freeswitch -> sipp
>>> >>
>>> >> Just 1 dialpeer with bridge action. No gateways. Just simple dialplan
>>> and 1 profile...
>>> >> Any advice?
>>> >>
>>> >> Thank you all
>>> >>
>>> >>
>>> >>
>>> >>
>>> >>
>>> >>
>>> _________________________________________________________________________
>>> >> Professional FreeSWITCH Consulting Services:
>>> >> consulting at freeswitch.org
>>> >> http://www.freeswitchsolutions.com
>>> >>
>>> >> Official FreeSWITCH Sites
>>> >> http://www.freeswitch.org
>>> >> http://confluence.freeswitch.org
>>> >> http://www.cluecon.com
>>> >>
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> > --
>>> > With kind regards
>>> > Peter Steinbach
>>> >
>>> > Telefaks Services GmbH
>>> > mailto:lists  (att) telefaks.de
>>> > Internet:www.telefaks.de
>>> >
>>> >
>>> >
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://confluence.freeswitch.org
>>> > http://www.cluecon.com
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>>
>>> --
>>> ------------------------------------------------------------
>>> Nathan Neulinger                       nneul at mst.edu
>>> Missouri S&T Information Technology    (573) 612-1412
>>> System Administrator - Architect
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> 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 bilaln018 at gmail.com  Thu Feb 11 17:54:05 2016
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Thu, 11 Feb 2016 19:54:05 +0500
Subject: [Freeswitch-users] [SDP Negotiation Error][Calling through
	JsSIP]
In-Reply-To: <CAEJMVkCrUZwSUhjM6s4+H5N28=i_0Qj+FoqjuAiDzYW0D1FCUA@mail.gmail.com>
References: <CADSVerCN05+5vvwvXKY=SrPhPwg8TKAWMUmXBv3P_y=tqmEr6A@mail.gmail.com>
	<CAEJMVkCrUZwSUhjM6s4+H5N28=i_0Qj+FoqjuAiDzYW0D1FCUA@mail.gmail.com>
Message-ID: <CADSVerAwUUMTpzAJLXc0_6YMyYS1PzusQHeyHe6tFLzoucWyXg@mail.gmail.com>

Thanks brain for reply but what i need to do, how can i get rid of this
warning.

many thanks

abbasi

On Thursday, February 11, 2016, Brian West <brian at freeswitch.org> wrote:

> NO candidate ACL defined, Defaulting to wan.auto
> sofia/internal/1001 at 192.241.213.201:7000 no suitable candidates found.
>
> On Thu, Feb 11, 2016 at 3:19 AM, Bilal Abbasi <bilaln018 at gmail.com
> <javascript:_e(%7B%7D,'cvml','bilaln018 at gmail.com');>> wrote:
>
>> Hi all,
>>
>> Currently i am facing an CODEC NEGOTIATION ERROR while calling through
>> JsSIP Caller 1001 Callie 1000,
>> Please view the logs,
>> https://pastebin.freeswitch.org/24550
>>
>> I have enabled PCMA and PCMU in my var.xml,
>>
>> Regards
>> Abbasi
>>
>> _________________________________________________________________________
>> 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
>
> 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 krice at freeswitch.org  Thu Feb 11 17:58:19 2016
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 11 Feb 2016 08:58:19 -0600
Subject: [Freeswitch-users] Jessie: Unable to locate package
In-Reply-To: <CCB34BD8-C5CA-47DF-B220-48079FEECEF3@interloop-software.de>
References: <CCB34BD8-C5CA-47DF-B220-48079FEECEF3@interloop-software.de>
Message-ID: <CAPcNkdd6De3ka6t1P4in9MWuTp6azFFxAaj2s4wg4SDYA6WYVA@mail.gmail.com>

are you sure you are using 64bit?

show us the contents of your apt sources.list.d file for freeswitch and
uname -a

On Thu, Feb 11, 2016 at 7:54 AM, Dominik Steinbrecher <
mailings at interloop-software.de> wrote:

> Hi,
>
> I want to install freeswitch 1.6 on a freshly installed Debian 8.3 jessie.
>
> I followed the guide from:
> https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie
>
> When I execute apt-get install I got the following errors:
>
> root at freeswitch:~# apt-get update && apt-get install -y freeswitch-all
> freeswitch-all-dbg gdb
> Hit http://security.debian.org jessie/updates InRelease
> Ign http://ftp.de.debian.org jessie InRelease
>
> Hit http://ftp.de.debian.org jessie-updates InRelease
>
> Hit http://ftp.de.debian.org jessie Release.gpg
>
> Hit http://security.debian.org jessie/updates/main Sources
>
> Hit http://ftp.de.debian.org jessie Release
>
> Hit http://security.debian.org jessie/updates/main i386 Packages
>
> Hit http://security.debian.org jessie/updates/main Translation-en
>
> Hit http://ftp.de.debian.org jessie-updates/main Sources
>
> Get:1 http://ftp.de.debian.org jessie-updates/main i386
> Packages/DiffIndex [367 B]
> Hit http://ftp.de.debian.org jessie-updates/main Translation-en
> Hit http://ftp.de.debian.org jessie/main Sources
> Hit http://ftp.de.debian.org jessie/main i386 Packages
> Hit http://ftp.de.debian.org jessie/main Translation-en
> Hit http://files.freeswitch.org jessie InRelease
> Hit http://files.freeswitch.org jessie/main i386 Packages
> Ign http://files.freeswitch.org jessie/main Translation-en_US
> Ign http://files.freeswitch.org jessie/main Translation-en
> Fetched 367 B in 3s (102 B/s)
> Reading package lists... Done
> Reading package lists... Done
> Building dependency tree
> Reading state information... Done
> E: Unable to locate package freeswitch-all
> E: Unable to locate package freeswitch-all-dbg
>
> I searched for the freeswitch-all package with aptitude but there?s no
> such package.
>
> Any ideas, what I am doing wrong? Is there anything I could try or check?
>
> The debian jessie 8.3 64bit is running as VM under VirtualBox. The host
> computer is a Mac Mini with OS X 10.11.1. It?s a freshly installed VM just
> for testing freeswitch.
>
> Thanks a lot
> Dominik
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 adnan.ahmed1 at gmail.com  Thu Feb 11 17:32:10 2016
From: adnan.ahmed1 at gmail.com (Adnan Ahmed)
Date: Thu, 11 Feb 2016 09:32:10 -0500
Subject: [Freeswitch-users] FS Bridged call, no RTP until DTMF pressed
Message-ID: <CAORkSaavPyN8hhd6hEu6oJS4tUHo=Ug+vzAqq=gy-r2CbrC97A@mail.gmail.com>

Hi,

I have a peculiar situation in which I'm hoping someone can help me out
with.  I have a Dahdi trunk coming into Asterisk (*), which then sends the
call directly to freeswitch (FS), FS will then bridge this incoming call to
a SIP device.  The problem i'm having is that when FS bridges the call
there is no media (or RTP packets) sent back to asterisk until I press a
dtmf key from the caller side.

The reason that * is there is due to the fact that mod_freeTDM for FS
wasn't able to configure the trunk parameters required to control the T1
(E&M with Feature Group B MF), with chan_dahdi in * i was able to set that
up with signalling=featb.  The dialplan in asterisk is as follows,

[from-pstn]
> exten => _X.,1,NoOp(Incoming DID matches as ${EXTEN})
> exten => _X.,n,Answer()
> exten => _X.,n,Set(CALLERID(all)="0000000000"<0000000000>)
> exten => _X.,n,Dial(SIP/freeswitch/1819${EXTEN:0:7},90,M(send-dtmf-1)r)
> exten => _X.,n,Hangup()
>


> [macro-send-dtmf-1]
> exten => s,1,SendDTMF(1)


 I tried sending a DTMF from astersk, and FS recognizes the DTMF, but still
no RTP until the key is physically pressed on the caller side.  The
asterisk dialplan is very simple, answer the incoming dahdi call and send
it to FS via SIP.  Once the DTMF is pressed, the audio is complete and no
issues anymore, so its not a routing, or firewall issue.  Both asterisk and
FS run on the same machine (* on port 5065, and FS on 5060).  Looking at
the tcpdump traces, there really is no RTP from FS until after the DTMF is
pressed, but the RTP from asterisk is always there.

I have the output of "sofia global siptrace on" at the following pastebin:
https://pastebin.freeswitch.org/24552

In that SIP trace you will see the call as follows,

Incoming call from *
bridge to SIP device
Failure to connect to SIP device
Forward call to voicemail
bridge to voicemail
connects to voicemail system
hangup

I can press the DTMF at any point once the first bridge is dialed and will
start hearing the audio from that point onwards ... in this case i pressed
the DTMF key 1 (you see it being recognized in the FS sip trace log).  It
makes no difference if I wait to press the DTMF till the second bridge or
after the second bridge connects.

I have even tried it with a sip device that answers on the first bridge
session, and its the same scenario:  no audio until dtmf is pressed, again
making no difference if its pressed right away or 10 seconds after the call
is connected and the other party can hear me but i don't hear them until i
press the dtmf.

Thanks,
Adnan.
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From anthony.minessale at gmail.com  Thu Feb 11 19:15:31 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Feb 2016 10:15:31 -0600
Subject: [Freeswitch-users] Does freeswitch forks his processes?
In-Reply-To: <CAMCT=3Q03rBbL6Gw2Jtkiyw_mNZbkVEtP43j1d8OgVfRHQ5HeA@mail.gmail.com>
References: <CAMCT=3R8CLCX2YBbNghAbr6NouK08=MpXUB2T0puoZ-rhJz-uQ@mail.gmail.com>
	<005d01d15db9$744a3bd0$5cdeb370$@botecomm.com>
	<56B0C6FE.9070507@telefaks.de> <56B0C9B7.8000106@mst.edu>
	<CAMCT=3QXHk9p1pHqywozuEiNA2zV-a8b9sYeuCLrpHvjgx05ag@mail.gmail.com>
	<CAEJMVkA6YoUwOmmsNy6P6rNVebczY=K6ww2-GnBuDopc+a_=UQ@mail.gmail.com>
	<CAMCT=3Q03rBbL6Gw2Jtkiyw_mNZbkVEtP43j1d8OgVfRHQ5HeA@mail.gmail.com>
Message-ID: <CAKbxfG9q-1j++=ckJ1YaL48aFS1u5rDJH0xTQQmJAJ-fwtiqAg@mail.gmail.com>

Without specific details about what the CPU is, how many there are and the
version of the OS and many other factors its not easy to answer you.
We frown on these load questions because, just like now, it results in
large threads and many people spending a lot of time trying to understand
the parameters.




On Thu, Feb 11, 2016 at 8:45 AM, Vladislav Ivanov <deforceczt at gmail.com>
wrote:

> I have changed it:
>  <param name="sessions-per-second" value="400"/>
>
> If there was limit - there would be no CPU hogging.
> Here is status command:
> FreeSWITCH (Version 1.6.6 git d2d0b32 2016-01-11 20:16:12Z 64bit) is ready
> 0 session(s) since startup
> 0 session(s) - peak 0, last 5min 0
> 0 session(s) per Sec out of max 400, peak 0, last 5min 0
> 10000 session(s) max
> min idle cpu 0.00/100.00
> Current Stack Size/Max 240K/8192K
>
> I can believe that i'm just DDoSing my box.
> I'm using simple uac/uas from sipp, and certain load is ok, but after that
> it goes straight down with increase of just 5 cps... I mean it is strange
> that with 20 cps it's 25% loaded and with 25 cps it's 100% loaded.
>
> 2016-02-11 16:38 GMT+02:00 Brian West <brian at freeswitch.org>:
>
>> I sense that your load testing method is flawed and you're just DDoSing
>> the box.
>>
>> On Thu, Feb 11, 2016 at 6:16 AM, Vladislav Ivanov <deforceczt at gmail.com>
>> wrote:
>>
>>> Hey guys,
>>>
>>> It is true, I haven't ran it as root, I fixed it.
>>> But I still have issues with passing more than 50cps, same is
>>> happening...
>>> 4 cores CPU - CPU is good, 80% idle, load average: 0.81, 0.98
>>> This at 25 cps (I bridge the call, so cps*2)
>>> Then I add 5 cps more:
>>> load average: 1061.40, 333.52
>>> And all OS start's lagging as hell, and i'm unable to find issue, have
>>> no idea what is happening...
>>>
>>> My settings:
>>> http://pastebin.com/62B45z4i
>>>
>>> Anything I'm missing? Really strange that CPU drops like that...
>>>
>>>
>>>
>>> 2016-02-02 17:22 GMT+02:00 Nathan Neulinger <nneul at mst.edu>:
>>>
>>>> "Run as" != "Start as"
>>>>
>>>> If you insist on not starting FS as root to let it change user, like
>>>> most other daemons/services, you'll have to jump
>>>> through a bunch of extra steps using file system capabilities to give
>>>> it the ability to set scheduler parameters/etc
>>>> that are restricted to root normally.
>>>>
>>>> -- Nathan
>>>>
>>>> On 02/02/2016 09:10 AM, Peter Steinbach wrote:
>>>> > I've just stumpled over this:
>>>> >  >Is FreeSWITCH starting with root permissions? It needs this in
>>>> order to use the FIFO scheduler and access realtime
>>>> > threads. If not started as root, this would explain your CPS
>>>> limitations.
>>>> >
>>>> > We like to run Freeswitch as a non privileged user, due to security
>>>> concerns. So there are drawbacks here compared to
>>>> > running FS as root? Can we somehow quantify the differences?
>>>> >
>>>> > Best regards
>>>> > Peter
>>>> >
>>>> >
>>>> > On 02/02/16 13:58, Bote Man wrote:
>>>> >>
>>>> >> Is FreeSWITCH starting with root permissions? It needs this in order
>>>> to use the FIFO scheduler and access realtime
>>>> >> threads. If not started as root, this would explain your CPS
>>>> limitations. There are also limits that can be set in the
>>>> >> config files.
>>>> >>
>>>> >> After it starts it drops privileges to those specified on the
>>>> command line with ?u and ?g switches.
>>>> >>
>>>> >> FreeSWITCH uses multi-threading. I do not know about htop, but maybe
>>>> it is showing the multiple threads?
>>>> >>
>>>> >> top ?H shows each thread.
>>>> >>
>>>> >> ---
>>>> >>
>>>> >> Bote
>>>> >>
>>>> >> FreeSWITCH Docs Janitor
>>>> >>
>>>> >> http://freeswitch.org/confluence
>>>> >>
>>>> >> *From:*Vladislav Ivanov
>>>> >> *Sent:* Tuesday, 02 February, 2016 07:09
>>>> >> *Subject:* [Freeswitch-users] Does freeswitch forks his processes?
>>>> >>
>>>> >> Hey guys,
>>>> >>
>>>> >> I have a question about freeswitch process/threading usage.
>>>> >> So far that I haven't noticed freeswitch to fork himself, I have
>>>> only 1 freeswitch instance.
>>>> >> http://i.imgur.com/bdbYOwp.png
>>>> >>
>>>> >> But then I found screenshot of htop with freeswitch and noticed that
>>>> there is multiple freeswitch processes being run:
>>>> >> http://i.imgur.com/VNpl55z.jpg
>>>> >>
>>>> >> I'm having issues with "loading" the freeswitch after 50 cps in any
>>>> cpu/ram configuration.
>>>> >> Be it physical or virtual environment I cant pass the 50 cps mark.
>>>> >> I have strange issue with CPU usage on same CPS:
>>>> >>
>>>> >> http://i.imgur.com/8BdQWVL.png
>>>> >> http://i.imgur.com/mWRnoGr.png
>>>> >>
>>>> >> I timeload test freeswitch with 50cps for 5+ hours, and seems like
>>>> there is some kind of leak somewhere.
>>>> >> I have tested configuration on:
>>>> >> Debian 8
>>>> >> 2 core/8 gb ram
>>>> >> 4 core/8 gb ram (graphs are from here)
>>>> >> 8 core/32 gb ram
>>>> >>
>>>> >> and in all the tests I were not able to send more than 50 cps
>>>> without CPU dropping to 0 with all system starting to
>>>> >> respond really laggy.
>>>> >>
>>>> >> Test is:
>>>> >> sipp -> freeswitch -> sipp
>>>> >>
>>>> >> Just 1 dialpeer with bridge action. No gateways. Just simple
>>>> dialplan and 1 profile...
>>>> >> Any advice?
>>>> >>
>>>> >> Thank you all
>>>> >>
>>>> >>
>>>> >>
>>>> >>
>>>> >>
>>>> >>
>>>> _________________________________________________________________________
>>>> >> Professional FreeSWITCH Consulting Services:
>>>> >> consulting at freeswitch.org
>>>> >> http://www.freeswitchsolutions.com
>>>> >>
>>>> >> Official FreeSWITCH Sites
>>>> >> http://www.freeswitch.org
>>>> >> http://confluence.freeswitch.org
>>>> >> http://www.cluecon.com
>>>> >>
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >
>>>> >
>>>> > --
>>>> > With kind regards
>>>> > Peter Steinbach
>>>> >
>>>> > Telefaks Services GmbH
>>>> > mailto:lists  (att) telefaks.de
>>>> > Internet:www.telefaks.de
>>>> >
>>>> >
>>>> >
>>>> >
>>>> _________________________________________________________________________
>>>> > Professional FreeSWITCH Consulting Services:
>>>> > consulting at freeswitch.org
>>>> > http://www.freeswitchsolutions.com
>>>> >
>>>> > Official FreeSWITCH Sites
>>>> > http://www.freeswitch.org
>>>> > http://confluence.freeswitch.org
>>>> > http://www.cluecon.com
>>>> >
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>>
>>>> --
>>>> ------------------------------------------------------------
>>>> Nathan Neulinger                       nneul at mst.edu
>>>> Missouri S&T Information Technology    (573) 612-1412
>>>> System Administrator - Architect
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>> 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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 gmaruzz at gmail.com  Thu Feb 11 19:18:57 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 11 Feb 2016 17:18:57 +0100
Subject: [Freeswitch-users] Does freeswitch forks his processes?
In-Reply-To: <CAKbxfG9q-1j++=ckJ1YaL48aFS1u5rDJH0xTQQmJAJ-fwtiqAg@mail.gmail.com>
References: <CAMCT=3R8CLCX2YBbNghAbr6NouK08=MpXUB2T0puoZ-rhJz-uQ@mail.gmail.com>
	<005d01d15db9$744a3bd0$5cdeb370$@botecomm.com>
	<56B0C6FE.9070507@telefaks.de> <56B0C9B7.8000106@mst.edu>
	<CAMCT=3QXHk9p1pHqywozuEiNA2zV-a8b9sYeuCLrpHvjgx05ag@mail.gmail.com>
	<CAEJMVkA6YoUwOmmsNy6P6rNVebczY=K6ww2-GnBuDopc+a_=UQ@mail.gmail.com>
	<CAMCT=3Q03rBbL6Gw2Jtkiyw_mNZbkVEtP43j1d8OgVfRHQ5HeA@mail.gmail.com>
	<CAKbxfG9q-1j++=ckJ1YaL48aFS1u5rDJH0xTQQmJAJ-fwtiqAg@mail.gmail.com>
Message-ID: <CALXCt0q==JCVkQLkFODD7FCxyk+osP7q1cfzVW9Jk7jMS9sYJQ@mail.gmail.com>

:))
guys like talk about fast cars and
:))


On Thu, Feb 11, 2016 at 5:15 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Without specific details about what the CPU is, how many there are and the
> version of the OS and many other factors its not easy to answer you.
> We frown on these load questions because, just like now, it results in
> large threads and many people spending a lot of time trying to understand
> the parameters.
>
>
>
>
> On Thu, Feb 11, 2016 at 8:45 AM, Vladislav Ivanov <deforceczt at gmail.com>
> wrote:
>
>> I have changed it:
>>  <param name="sessions-per-second" value="400"/>
>>
>> If there was limit - there would be no CPU hogging.
>> Here is status command:
>> FreeSWITCH (Version 1.6.6 git d2d0b32 2016-01-11 20:16:12Z 64bit) is ready
>> 0 session(s) since startup
>> 0 session(s) - peak 0, last 5min 0
>> 0 session(s) per Sec out of max 400, peak 0, last 5min 0
>> 10000 session(s) max
>> min idle cpu 0.00/100.00
>> Current Stack Size/Max 240K/8192K
>>
>> I can believe that i'm just DDoSing my box.
>> I'm using simple uac/uas from sipp, and certain load is ok, but after
>> that it goes straight down with increase of just 5 cps... I mean it is
>> strange that with 20 cps it's 25% loaded and with 25 cps it's 100% loaded.
>>
>> 2016-02-11 16:38 GMT+02:00 Brian West <brian at freeswitch.org>:
>>
>>> I sense that your load testing method is flawed and you're just DDoSing
>>> the box.
>>>
>>> On Thu, Feb 11, 2016 at 6:16 AM, Vladislav Ivanov <deforceczt at gmail.com>
>>> wrote:
>>>
>>>> Hey guys,
>>>>
>>>> It is true, I haven't ran it as root, I fixed it.
>>>> But I still have issues with passing more than 50cps, same is
>>>> happening...
>>>> 4 cores CPU - CPU is good, 80% idle, load average: 0.81, 0.98
>>>> This at 25 cps (I bridge the call, so cps*2)
>>>> Then I add 5 cps more:
>>>> load average: 1061.40, 333.52
>>>> And all OS start's lagging as hell, and i'm unable to find issue, have
>>>> no idea what is happening...
>>>>
>>>> My settings:
>>>> http://pastebin.com/62B45z4i
>>>>
>>>> Anything I'm missing? Really strange that CPU drops like that...
>>>>
>>>>
>>>>
>>>> 2016-02-02 17:22 GMT+02:00 Nathan Neulinger <nneul at mst.edu>:
>>>>
>>>>> "Run as" != "Start as"
>>>>>
>>>>> If you insist on not starting FS as root to let it change user, like
>>>>> most other daemons/services, you'll have to jump
>>>>> through a bunch of extra steps using file system capabilities to give
>>>>> it the ability to set scheduler parameters/etc
>>>>> that are restricted to root normally.
>>>>>
>>>>> -- Nathan
>>>>>
>>>>> On 02/02/2016 09:10 AM, Peter Steinbach wrote:
>>>>> > I've just stumpled over this:
>>>>> >  >Is FreeSWITCH starting with root permissions? It needs this in
>>>>> order to use the FIFO scheduler and access realtime
>>>>> > threads. If not started as root, this would explain your CPS
>>>>> limitations.
>>>>> >
>>>>> > We like to run Freeswitch as a non privileged user, due to security
>>>>> concerns. So there are drawbacks here compared to
>>>>> > running FS as root? Can we somehow quantify the differences?
>>>>> >
>>>>> > Best regards
>>>>> > Peter
>>>>> >
>>>>> >
>>>>> > On 02/02/16 13:58, Bote Man wrote:
>>>>> >>
>>>>> >> Is FreeSWITCH starting with root permissions? It needs this in
>>>>> order to use the FIFO scheduler and access realtime
>>>>> >> threads. If not started as root, this would explain your CPS
>>>>> limitations. There are also limits that can be set in the
>>>>> >> config files.
>>>>> >>
>>>>> >> After it starts it drops privileges to those specified on the
>>>>> command line with ?u and ?g switches.
>>>>> >>
>>>>> >> FreeSWITCH uses multi-threading. I do not know about htop, but
>>>>> maybe it is showing the multiple threads?
>>>>> >>
>>>>> >> top ?H shows each thread.
>>>>> >>
>>>>> >> ---
>>>>> >>
>>>>> >> Bote
>>>>> >>
>>>>> >> FreeSWITCH Docs Janitor
>>>>> >>
>>>>> >> http://freeswitch.org/confluence
>>>>> >>
>>>>> >> *From:*Vladislav Ivanov
>>>>> >> *Sent:* Tuesday, 02 February, 2016 07:09
>>>>> >> *Subject:* [Freeswitch-users] Does freeswitch forks his processes?
>>>>> >>
>>>>> >> Hey guys,
>>>>> >>
>>>>> >> I have a question about freeswitch process/threading usage.
>>>>> >> So far that I haven't noticed freeswitch to fork himself, I have
>>>>> only 1 freeswitch instance.
>>>>> >> http://i.imgur.com/bdbYOwp.png
>>>>> >>
>>>>> >> But then I found screenshot of htop with freeswitch and noticed
>>>>> that there is multiple freeswitch processes being run:
>>>>> >> http://i.imgur.com/VNpl55z.jpg
>>>>> >>
>>>>> >> I'm having issues with "loading" the freeswitch after 50 cps in any
>>>>> cpu/ram configuration.
>>>>> >> Be it physical or virtual environment I cant pass the 50 cps mark.
>>>>> >> I have strange issue with CPU usage on same CPS:
>>>>> >>
>>>>> >> http://i.imgur.com/8BdQWVL.png
>>>>> >> http://i.imgur.com/mWRnoGr.png
>>>>> >>
>>>>> >> I timeload test freeswitch with 50cps for 5+ hours, and seems like
>>>>> there is some kind of leak somewhere.
>>>>> >> I have tested configuration on:
>>>>> >> Debian 8
>>>>> >> 2 core/8 gb ram
>>>>> >> 4 core/8 gb ram (graphs are from here)
>>>>> >> 8 core/32 gb ram
>>>>> >>
>>>>> >> and in all the tests I were not able to send more than 50 cps
>>>>> without CPU dropping to 0 with all system starting to
>>>>> >> respond really laggy.
>>>>> >>
>>>>> >> Test is:
>>>>> >> sipp -> freeswitch -> sipp
>>>>> >>
>>>>> >> Just 1 dialpeer with bridge action. No gateways. Just simple
>>>>> dialplan and 1 profile...
>>>>> >> Any advice?
>>>>> >>
>>>>> >> Thank you all
>>>>> >>
>>>>> >>
>>>>> >>
>>>>> >>
>>>>> >>
>>>>> >>
>>>>> _________________________________________________________________________
>>>>> >> Professional FreeSWITCH Consulting Services:
>>>>> >> consulting at freeswitch.org
>>>>> >> http://www.freeswitchsolutions.com
>>>>> >>
>>>>> >> Official FreeSWITCH Sites
>>>>> >> http://www.freeswitch.org
>>>>> >> http://confluence.freeswitch.org
>>>>> >> http://www.cluecon.com
>>>>> >>
>>>>> >> FreeSWITCH-users mailing list
>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> http://www.freeswitch.org
>>>>> >
>>>>> >
>>>>> > --
>>>>> > With kind regards
>>>>> > Peter Steinbach
>>>>> >
>>>>> > Telefaks Services GmbH
>>>>> > mailto:lists  (att) telefaks.de
>>>>> > Internet:www.telefaks.de
>>>>> >
>>>>> >
>>>>> >
>>>>> >
>>>>> _________________________________________________________________________
>>>>> > Professional FreeSWITCH Consulting Services:
>>>>> > consulting at freeswitch.org
>>>>> > http://www.freeswitchsolutions.com
>>>>> >
>>>>> > Official FreeSWITCH Sites
>>>>> > http://www.freeswitch.org
>>>>> > http://confluence.freeswitch.org
>>>>> > http://www.cluecon.com
>>>>> >
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> > UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > http://www.freeswitch.org
>>>>> >
>>>>>
>>>>> --
>>>>> ------------------------------------------------------------
>>>>> Nathan Neulinger                       nneul at mst.edu
>>>>> Missouri S&T Information Technology    (573) 612-1412
>>>>> System Administrator - Architect
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>>
>>> 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
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From mike at jerris.com  Thu Feb 11 19:21:11 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Feb 2016 10:21:11 -0600
Subject: [Freeswitch-users] Two Users Registered ..But calls only going
	one way
In-Reply-To: <CAK9Ws2dOCsLXVS=JPzj4yW-RV_SffNYrWYpoMn=AQy8x_x5gKw@mail.gmail.com>
References: <CAK9Ws2dOCsLXVS=JPzj4yW-RV_SffNYrWYpoMn=AQy8x_x5gKw@mail.gmail.com>
Message-ID: <792D69F0-C82B-4DE7-B7D6-58F19FADC60C@jerris.com>

Sip trace would help... is call forwarding turned on on the phone?

> On Feb 11, 2016, at 3:46 AM, David Wafula <davidwaf at gmail.com> wrote:
> 
> Hi all,
> I have two users who registered in the same domain: user A and B.
> A can call B just fine. When B tries to call A, there is silence (no ringback)..then after sometime the call goes into voice mail. A never receives the call.  Please see the call trace here:
> 
> 
> http://pastebin.com/gWrrS4zw <http://pastebin.com/gWrrS4zw>
> 
> Am not sure what is causing it.

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From shafeeq.v at gmail.com  Thu Feb 11 19:28:45 2016
From: shafeeq.v at gmail.com (mohammed shafeeque)
Date: Thu, 11 Feb 2016 21:58:45 +0530
Subject: [Freeswitch-users] Oneway audio issues in freeswitch
In-Reply-To: <CAFLo2QzdR1YBLv6-y5hs=juyE5N2uZw+N5dCiyq4S3Hbgkb2eg@mail.gmail.com>
References: <CANw+5nWmd_j=eUz7OGoH=5Wb3g43VOS_XF_dZVkuFRMRUtfZ2Q@mail.gmail.com>
	<CABMX_w_NWvQ5ZUdxtYHnua4FD9i4hYDt3YeP-X+43XYi4cb=8A@mail.gmail.com>
	<CAFLo2QzdR1YBLv6-y5hs=juyE5N2uZw+N5dCiyq4S3Hbgkb2eg@mail.gmail.com>
Message-ID: <CANw+5nUWV3dmJGUtDgLBSaUuGufcXskoYNyj6FJcmd+yRs184g@mail.gmail.com>

Surprised that no one else experienced this problem. Can anyone give any
hint. Really Dont want to move back to 1.4.x

On Thu, Feb 11, 2016 at 7:44 AM, Luis Daniel Lucio Quiroz <
luis.daniel.lucio at gmail.com> wrote:

> As a rule of dumb, try turning on rport
> Le 10 f?vr. 2016 8:55 PM, "?talo Rossi" <italo at freeswitch.org> a ?crit :
>
>> You need to look at the sip signaling to see what's going on
>>
>> On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque <shafeeq.v at gmail.com>
>> wrote:
>>
>>> Hello All
>>>
>>> We are getting one way audio issues with some softphones and grandstream
>>> phones behind nat registerd to our freeswitch server.
>>>
>>> Here is scenario:
>>> Grandstream call any extensions (one way audio)
>>> Any extension call Grandstream ( Audio works just fine)
>>>
>>> We have tried multiple softphones and the result is same.
>>>
>>> Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF
>>> issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started
>>> with an upgrade to freeswitch.
>>>
>>> Any help or hint will be much appreciated.
>>>
>>> Thank you,
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> ?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
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 stefano.favaro at edistar.com  Thu Feb 11 19:33:10 2016
From: stefano.favaro at edistar.com (Stefano Favaro)
Date: Thu, 11 Feb 2016 17:33:10 +0100 (CET)
Subject: [Freeswitch-users] Problem with mod_spy
In-Reply-To: <374662561.6043.1455208118323.JavaMail.root@mailserver.edistar.com>
Message-ID: <965414335.6054.1455208390800.JavaMail.root@mailserver.edistar.com>

Hello,

 I have a problem with the mod_spy module. 
It seems that it just plays music and do not actually spy.
I want to dial an extension with the number of the user I want to spy and wait for a call in or out for that user. 
Mod_spy application plays music but when a call is handled by the user I can't hear it continues to play the music on hold audio file. 

This is the dialplan:

<extension name="eavesdrop"> 
<condition field="destination_number" expression="^88(.*)$|^\*0(.*)$"> 
<action application="answer"/> 
<action application="userspy" data="$1 at myserver"/> 
</condition> 
</extension> 
If I dial 881000, for example, It means I want to spy on user 1000.
I have in and out calls from user 1000 but I can't hear. 
userspy_show in fs_cli, I get : 
1000 at myserver : e9165eaf-e8d7-40ad-a8cd-d2963f363684

1 total spy 

I have FreeSwitch version FreeSWITCH Version 1.4.26-37~64bit (-37 64bit) 

SF. 
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From mailings at interloop-software.de  Thu Feb 11 19:35:33 2016
From: mailings at interloop-software.de (Dominik Steinbrecher)
Date: Thu, 11 Feb 2016 17:35:33 +0100
Subject: [Freeswitch-users] Jessie: Unable to locate package
In-Reply-To: <CAPcNkdd6De3ka6t1P4in9MWuTp6azFFxAaj2s4wg4SDYA6WYVA@mail.gmail.com>
References: <CCB34BD8-C5CA-47DF-B220-48079FEECEF3@interloop-software.de>
	<CAPcNkdd6De3ka6t1P4in9MWuTp6azFFxAaj2s4wg4SDYA6WYVA@mail.gmail.com>
Message-ID: <2D6C19DC-5C1F-4590-B7A4-1D1708F5FB1C@interloop-software.de>

Thanks for your answers and help.

> show us the contents of your apt sources.list.d file for freeswitch and uname -a


root at freeswitch:~# cat /etc/apt/sources.list.d/freeswitch.list 
deb http://files.freeswitch.org/repo/deb/freeswitch-1.6/ jessie main

root at freeswitch:~# uname -a
Linux freeswitch 3.16.0-4-686-pae #1 SMP Debian 3.16.7-ckt20-1+deb8u3 (2016-01-17) i686 GNU/Linux

> are you sure you are using 64bit? 

I thought that I was running a 64bit OS, but it looks like I was wrong. As I showed above, uname -a said i686 so its 32bit. 

Sorry, for the wrong information in my first post. Looks like that I was choosing the wrong netinst-image for download after selecting 64bit OS in VirtualBox.

I just installed a new vm with arch amd64 and everything seems to be fine. The freeswitch installation just finished without any errors. 

So, thank you a lot for your help
Dominik 


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From krice at freeswitch.org  Thu Feb 11 19:36:37 2016
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 11 Feb 2016 10:36:37 -0600
Subject: [Freeswitch-users] Oneway audio issues in freeswitch
In-Reply-To: <CANw+5nUWV3dmJGUtDgLBSaUuGufcXskoYNyj6FJcmd+yRs184g@mail.gmail.com>
References: <CANw+5nWmd_j=eUz7OGoH=5Wb3g43VOS_XF_dZVkuFRMRUtfZ2Q@mail.gmail.com>
	<CABMX_w_NWvQ5ZUdxtYHnua4FD9i4hYDt3YeP-X+43XYi4cb=8A@mail.gmail.com>
	<CAFLo2QzdR1YBLv6-y5hs=juyE5N2uZw+N5dCiyq4S3Hbgkb2eg@mail.gmail.com>
	<CANw+5nUWV3dmJGUtDgLBSaUuGufcXskoYNyj6FJcmd+yRs184g@mail.gmail.com>
Message-ID: <CAPcNkdfppepyyC+OLTr4OFyhFWhP9MjWR0WUsraU-ABrcJLY1Q@mail.gmail.com>

without logs of a call doing this at debug level with a complete unmolested
sip trace in line its a little hard to speculate whats going on here

On Thu, Feb 11, 2016 at 10:28 AM, mohammed shafeeque <shafeeq.v at gmail.com>
wrote:

> Surprised that no one else experienced this problem. Can anyone give any
> hint. Really Dont want to move back to 1.4.x
>
> On Thu, Feb 11, 2016 at 7:44 AM, Luis Daniel Lucio Quiroz <
> luis.daniel.lucio at gmail.com> wrote:
>
>> As a rule of dumb, try turning on rport
>> Le 10 f?vr. 2016 8:55 PM, "?talo Rossi" <italo at freeswitch.org> a ?crit :
>>
>>> You need to look at the sip signaling to see what's going on
>>>
>>> On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque <shafeeq.v at gmail.com
>>> > wrote:
>>>
>>>> Hello All
>>>>
>>>> We are getting one way audio issues with some softphones and
>>>> grandstream phones behind nat registerd to our freeswitch server.
>>>>
>>>> Here is scenario:
>>>> Grandstream call any extensions (one way audio)
>>>> Any extension call Grandstream ( Audio works just fine)
>>>>
>>>> We have tried multiple softphones and the result is same.
>>>>
>>>> Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF
>>>> issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started
>>>> with an upgrade to freeswitch.
>>>>
>>>> Any help or hint will be much appreciated.
>>>>
>>>> Thank you,
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> ?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
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Thu Feb 11 19:53:55 2016
From: lists at kavun.ch (Emrah)
Date: Thu, 11 Feb 2016 17:53:55 +0100
Subject: [Freeswitch-users] High availability on different networks
Message-ID: <78429801-FCA4-4BA9-8EA6-64A1BD7ECA28@kavun.ch>

Hi list,
I?m writing to gather your thoughts and suggestions on how to have a high availability FS setup on different networks.

I am trying to achieve the following:
 - Load balance FreeSWITCH instances on 2 or more servers, possibly in different countries.
- Shared user directory and dialplan, but I?m not sure if shared registrations would make sense. 
- If a server goes down, the phone should register on the alternative servers. Obviously we can?t keep calls up.

I?m obviously not the first one out there doing this. I?m trying to learn from those who?ve come up with reliable solutions.

I?ve tried sharing a registration table among multiple FS instances. But it was a beginners mistake. Even with the right path to reach the client, only the invites sent from the server used by the phone would be processed. 
If my phone registers on server A, then server A shares the info with server B, server B knows how to contact the phone but it won?t be able to. Supposedly because of NAT issues.

I am aiming for fully independent FS instances that can back each other up and be used independently. I am guessing this would require some sort of SBC or external registrar server with a Kamailio or Repro. 

Anyway just trying to spark the conversation around this subject and hopefully we can come up with a formula that can help many with their FS deployments. My provider?s network just went all down in IPv4 and HA behind the same provider proved to be useless.

Best,
E

From jjserranor at gmail.com  Thu Feb 11 19:59:25 2016
From: jjserranor at gmail.com (Jose Serrano)
Date: Thu, 11 Feb 2016 17:59:25 +0100
Subject: [Freeswitch-users] P-asserted-identity
In-Reply-To: <CAFiqYu=gYbYsqkvySTJ2qhRXFjWvVFS1r+-zDe8om7S5D8yzGg@mail.gmail.com>
References: <CAArQ-M=aa0rN-pXKe0OXfYy8bQs9qoNVS51TPopu37J0_e2XkA@mail.gmail.com>
	<CAFiqYu=gYbYsqkvySTJ2qhRXFjWvVFS1r+-zDe8om7S5D8yzGg@mail.gmail.com>
Message-ID: <CAArQ-M=hV9-o2cq8w+6HG-hYhRC4SbdJKwcQQARSS+X2zh+Xzw@mail.gmail.com>

Hello

Sorry. I made a mistake checking what "royj at yandex.ru" told me.
I can confirm that using <param name="caller-id-type" value="pid"/> works
fine.

Nevertheless  "royj at yandex.ru" told me to apply it  in the incoming
profile, but it works when I have applied in the outbound profile.

Thanks everybody. I really apreciate your help

SOLVED

2016-02-11 11:18 GMT+01:00 Steven Ayre <steveayre at gmail.com>:

> The only thing that works is to set up the "sip_cid_type" variable in the
>> dialplan like this:
>
> <action application="bridge" data="{sip_cid_type=pid}sofia/
>> gateway/Mygateway/$1"/>
>
>
> Try setting sip_cid_type=pid as a variable within the gateway definition,
> so it's automatically set on every outbound call through that gateway.
>
>
> https://freeswitch.org/confluence/display/FREESWITCH/Gateways+Configuration#GatewaysConfiguration-Variables
>
> On 10 February 2016 at 10:31, Jose Serrano <jjserranor at gmail.com> wrote:
>
>> Hello.
>>
>> My freeswitch  by default replace the P-asserted-identity  by
>> Remote-Party-ID when routing calls to the gateway.
>> I want to send the P-asserted-identity and not the Remote-Party-ID and
>> for that I tried the following:
>>
>> I have configured  in the outbound gateway definition the following
>> parameters:
>> <param name="caller-id-type" value="pid"/> or
>> <param name="cid-type" value="pid"/>
>>  but te behavior is the same.
>>
>> The only thing that works is to set up the "sip_cid_type" variable in the
>> dialplan like this:
>> <action application="bridge"
>> data="{sip_cid_type=pid}sofia/gateway/Mygateway/$1"/>
>>
>> Anyone  knows how I can send the P-Asserted-identity without having to
>> modify all my dial plan adding the sip_cid_type?
>>
>> Thanks in avanced
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 italo at freeswitch.org  Thu Feb 11 20:52:46 2016
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Thu, 11 Feb 2016 14:52:46 -0300
Subject: [Freeswitch-users] Problem with mod_spy
In-Reply-To: <965414335.6054.1455208390800.JavaMail.root@mailserver.edistar.com>
References: <374662561.6043.1455208118323.JavaMail.root@mailserver.edistar.com>
	<965414335.6054.1455208390800.JavaMail.root@mailserver.edistar.com>
Message-ID: <CABMX_w-SYNKF4_E5Ui+ou0y-qvM8awG0nodCvTgKqRZRtbJ=cw@mail.gmail.com>

Stefano,

Can you post your debug logs (/log 7)?

Use https://pastebin.freeswitch.org/

On Thu, Feb 11, 2016 at 1:33 PM, Stefano Favaro <stefano.favaro at edistar.com>
wrote:

> Hello,
>
>  I have a problem with the mod_spy module.
> It seems that it just plays music and do not actually spy.
> I want to dial an extension with the number of the user I want to spy and wait for a call in or out for that user.
> Mod_spy application plays music but when a call is handled by the user I can't hear it continues to play the music on hold audio file.
>
> This is the dialplan:
>
> <extension name="eavesdrop">
>   <condition field="destination_number" expression="^88(.*)$|^\*0(.*)$">
>     <action application="answer"/>
>     <action application="userspy" data="$1 at myserver"/>
>   </condition>
> </extension>
>
>
> If I dial 881000, for example, It means I want to spy on user 1000.
> I have in and out calls from user 1000 but I can't hear.
> userspy_show in fs_cli, I get :
> 1000 at myserver : e9165eaf-e8d7-40ad-a8cd-d2963f363684
>
> 1 total spy
>
> I have FreeSwitch version FreeSWITCH Version 1.4.26-37~64bit (-37 64bit)
>
> SF.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From ahabiba at gmail.com  Thu Feb 11 21:05:58 2016
From: ahabiba at gmail.com (Ahmed Habiba)
Date: Thu, 11 Feb 2016 21:05:58 +0300
Subject: [Freeswitch-users] Two Users Registered ..But calls only going
	one way
In-Reply-To: <mailman.2084.1455208631.678.freeswitch-users@lists.freeswitch.org>
References: <mailman.2084.1455208631.678.freeswitch-users@lists.freeswitch.org>
Message-ID: <FD5DFF5D-1668-46AD-A489-D42A6BD23677@gmail.com>

Are you using TLS? is GS configures with nat configuration correctly?

> 
> From: Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>>
> Subject: Re: [Freeswitch-users] Two Users Registered ..But calls only going one way
> Date: February 11, 2016 at 7:21:11 PM GMT+3
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> 
> 
> Sip trace would help... is call forwarding turned on on the phone?
> 
>> On Feb 11, 2016, at 3:46 AM, David Wafula <davidwaf at gmail.com <mailto:davidwaf at gmail.com>> wrote:
>> 
>> Hi all,
>> I have two users who registered in the same domain: user A and B.
>> A can call B just fine. When B tries to call A, there is silence (no ringback)..then after sometime the call goes into voice mail. A never receives the call.  Please see the call trace here:
>> 
>> 
>> http://pastebin.com/gWrrS4zw <http://pastebin.com/gWrrS4zw>
>> 
>> Am not sure what is causing it.
> 
> 
> 
> 
> From: mohammed shafeeque <shafeeq.v at gmail.com <mailto:shafeeq.v at gmail.com>>
> Subject: Re: [Freeswitch-users] Oneway audio issues in freeswitch
> Date: February 11, 2016 at 7:28:45 PM GMT+3
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> 
> 
> Surprised that no one else experienced this problem. Can anyone give any hint. Really Dont want to move back to 1.4.x
> 
> On Thu, Feb 11, 2016 at 7:44 AM, Luis Daniel Lucio Quiroz <luis.daniel.lucio at gmail.com <mailto:luis.daniel.lucio at gmail.com>> wrote:
> As a rule of dumb, try turning on rport
> Le 10 f?vr. 2016 8:55 PM, "?talo Rossi" <italo at freeswitch.org <mailto:italo at freeswitch.org>> a ?crit :
> You need to look at the sip signaling to see what's going on
> 
> On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque <shafeeq.v at gmail.com <mailto:shafeeq.v at gmail.com>> wrote:
> Hello All
> 
> We are getting one way audio issues with some softphones and grandstream phones behind nat registerd to our freeswitch server. 
> 
> Here is scenario: 
> Grandstream call any extensions (one way audio) 
> Any extension call Grandstream ( Audio works just fine)
> 
> We have tried multiple softphones and the result is same. 
> 
> Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started with an upgrade to freeswitch. 
> 
> Any help or hint will be much appreciated. 
> 
> Thank you, 
> 
> 
> _________________________________________________________________________
> 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/>
> 
> 
> 
> -- 
> ?talo Rossi
> italo at freeswitch.org <mailto:italo at freeswitch.org>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> 

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From ssinyagin at gmail.com  Thu Feb 11 22:00:25 2016
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Thu, 11 Feb 2016 20:00:25 +0100
Subject: [Freeswitch-users] High availability on different networks
In-Reply-To: <78429801-FCA4-4BA9-8EA6-64A1BD7ECA28@kavun.ch>
References: <78429801-FCA4-4BA9-8EA6-64A1BD7ECA28@kavun.ch>
Message-ID: <CACOYK=gUYrGPouHq4voAygR7ctBOc6sdwBrAF8DWEeW6XkXS9A@mail.gmail.com>

hi Emrah and all,

it's the first time I actually searched for it, but there are hosting
offers with anycast IP routing. It means, you have multiple servers in
various locations, and they share the same service IP address. The
clients connect to the nearest server, which is determined by standard
BGP routing. You are still limited to a single global hosting
provider, but you benefit from its redundant network and geographical
distribution.

In case of anycast addressing, incoming connections will be served
easily. But the outgoing connections are rather tricky: you will need
to bring the outbound call to the physical server where the user has
registered, and initiate the connection from its anycast address. So,
you can share and replicate the registration database, but you need to
send the outbound call to the server which accepted the registration.
I guess you should be able to retrieve this information from the
registration database. This needs to be looked in details.

Google for anycast server hosting, and there are at least 3 providers
offering virtual hosts, and OVH is offering physical hosts as well. I
guess there are more providers with similar offerings.


Without anycast, you would need to use redundant registrars sharing
the same service IP address -- for example, Digitalocean offers such
service within any single datacenter.

Having multiple registrars with different IP addresses is also
possible, but then you depend on the way how each particular SIP
client handles multiple IP addresses after resolving the domain name.
Some of them may get stuck to a single address, even if it's not
responding.


cheers,
stanislav

On Thu, Feb 11, 2016 at 5:53 PM, Emrah <lists at kavun.ch> wrote:
> Hi list,
> I?m writing to gather your thoughts and suggestions on how to have a high availability FS setup on different networks.
>
> I am trying to achieve the following:
>  - Load balance FreeSWITCH instances on 2 or more servers, possibly in different countries.
> - Shared user directory and dialplan, but I?m not sure if shared registrations would make sense.
> - If a server goes down, the phone should register on the alternative servers. Obviously we can?t keep calls up.
>
> I?m obviously not the first one out there doing this. I?m trying to learn from those who?ve come up with reliable solutions.
>
> I?ve tried sharing a registration table among multiple FS instances. But it was a beginners mistake. Even with the right path to reach the client, only the invites sent from the server used by the phone would be processed.
> If my phone registers on server A, then server A shares the info with server B, server B knows how to contact the phone but it won?t be able to. Supposedly because of NAT issues.
>
> I am aiming for fully independent FS instances that can back each other up and be used independently. I am guessing this would require some sort of SBC or external registrar server with a Kamailio or Repro.
>
> Anyway just trying to spark the conversation around this subject and hopefully we can come up with a formula that can help many with their FS deployments. My provider?s network just went all down in IPv4 and HA behind the same provider proved to be useless.
>
> Best,
> E
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 vipkilla at gmail.com  Thu Feb 11 22:27:27 2016
From: vipkilla at gmail.com (Vik Killa)
Date: Thu, 11 Feb 2016 14:27:27 -0500
Subject: [Freeswitch-users] High availability on different networks
In-Reply-To: <CACOYK=gUYrGPouHq4voAygR7ctBOc6sdwBrAF8DWEeW6XkXS9A@mail.gmail.com>
References: <78429801-FCA4-4BA9-8EA6-64A1BD7ECA28@kavun.ch>
	<CACOYK=gUYrGPouHq4voAygR7ctBOc6sdwBrAF8DWEeW6XkXS9A@mail.gmail.com>
Message-ID: <CAC-LwPOdcwKvUYiR=1RzmrfdS9c_HHHnRg9QhFpsHU18coBjvw@mail.gmail.com>

What about DNS SRV?

On Thu, Feb 11, 2016 at 2:00 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> hi Emrah and all,
>
> it's the first time I actually searched for it, but there are hosting
> offers with anycast IP routing. It means, you have multiple servers in
> various locations, and they share the same service IP address. The
> clients connect to the nearest server, which is determined by standard
> BGP routing. You are still limited to a single global hosting
> provider, but you benefit from its redundant network and geographical
> distribution.
>
> In case of anycast addressing, incoming connections will be served
> easily. But the outgoing connections are rather tricky: you will need
> to bring the outbound call to the physical server where the user has
> registered, and initiate the connection from its anycast address. So,
> you can share and replicate the registration database, but you need to
> send the outbound call to the server which accepted the registration.
> I guess you should be able to retrieve this information from the
> registration database. This needs to be looked in details.
>
> Google for anycast server hosting, and there are at least 3 providers
> offering virtual hosts, and OVH is offering physical hosts as well. I
> guess there are more providers with similar offerings.
>
>
> Without anycast, you would need to use redundant registrars sharing
> the same service IP address -- for example, Digitalocean offers such
> service within any single datacenter.
>
> Having multiple registrars with different IP addresses is also
> possible, but then you depend on the way how each particular SIP
> client handles multiple IP addresses after resolving the domain name.
> Some of them may get stuck to a single address, even if it's not
> responding.
>
>
> cheers,
> stanislav
>
> On Thu, Feb 11, 2016 at 5:53 PM, Emrah <lists at kavun.ch> wrote:
> > Hi list,
> > I?m writing to gather your thoughts and suggestions on how to have a
> high availability FS setup on different networks.
> >
> > I am trying to achieve the following:
> >  - Load balance FreeSWITCH instances on 2 or more servers, possibly in
> different countries.
> > - Shared user directory and dialplan, but I?m not sure if shared
> registrations would make sense.
> > - If a server goes down, the phone should register on the alternative
> servers. Obviously we can?t keep calls up.
> >
> > I?m obviously not the first one out there doing this. I?m trying to
> learn from those who?ve come up with reliable solutions.
> >
> > I?ve tried sharing a registration table among multiple FS instances. But
> it was a beginners mistake. Even with the right path to reach the client,
> only the invites sent from the server used by the phone would be processed.
> > If my phone registers on server A, then server A shares the info with
> server B, server B knows how to contact the phone but it won?t be able to.
> Supposedly because of NAT issues.
> >
> > I am aiming for fully independent FS instances that can back each other
> up and be used independently. I am guessing this would require some sort of
> SBC or external registrar server with a Kamailio or Repro.
> >
> > Anyway just trying to spark the conversation around this subject and
> hopefully we can come up with a formula that can help many with their FS
> deployments. My provider?s network just went all down in IPv4 and HA behind
> the same provider proved to be useless.
> >
> > Best,
> > E
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Thu Feb 11 23:16:45 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 11 Feb 2016 23:16:45 +0300
Subject: [Freeswitch-users] High availability on different networks
In-Reply-To: <CAC-LwPOdcwKvUYiR=1RzmrfdS9c_HHHnRg9QhFpsHU18coBjvw@mail.gmail.com>
References: <78429801-FCA4-4BA9-8EA6-64A1BD7ECA28@kavun.ch>
	<CACOYK=gUYrGPouHq4voAygR7ctBOc6sdwBrAF8DWEeW6XkXS9A@mail.gmail.com>
	<CAC-LwPOdcwKvUYiR=1RzmrfdS9c_HHHnRg9QhFpsHU18coBjvw@mail.gmail.com>
Message-ID: <CAHtxdDdp=czW4sHONLs8mGpzYgLuNg1K4GSp2uKF_LA9_FgsUw@mail.gmail.com>

I use A record updated to point active FS servers in geo-distributed
cluster.
I DNS server i can recommend use Amazon Route53

On Thu, Feb 11, 2016 at 10:27 PM, Vik Killa <vipkilla at gmail.com> wrote:

> What about DNS SRV?
>
> On Thu, Feb 11, 2016 at 2:00 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
>
>> hi Emrah and all,
>>
>> it's the first time I actually searched for it, but there are hosting
>> offers with anycast IP routing. It means, you have multiple servers in
>> various locations, and they share the same service IP address. The
>> clients connect to the nearest server, which is determined by standard
>> BGP routing. You are still limited to a single global hosting
>> provider, but you benefit from its redundant network and geographical
>> distribution.
>>
>> In case of anycast addressing, incoming connections will be served
>> easily. But the outgoing connections are rather tricky: you will need
>> to bring the outbound call to the physical server where the user has
>> registered, and initiate the connection from its anycast address. So,
>> you can share and replicate the registration database, but you need to
>> send the outbound call to the server which accepted the registration.
>> I guess you should be able to retrieve this information from the
>> registration database. This needs to be looked in details.
>>
>> Google for anycast server hosting, and there are at least 3 providers
>> offering virtual hosts, and OVH is offering physical hosts as well. I
>> guess there are more providers with similar offerings.
>>
>>
>> Without anycast, you would need to use redundant registrars sharing
>> the same service IP address -- for example, Digitalocean offers such
>> service within any single datacenter.
>>
>> Having multiple registrars with different IP addresses is also
>> possible, but then you depend on the way how each particular SIP
>> client handles multiple IP addresses after resolving the domain name.
>> Some of them may get stuck to a single address, even if it's not
>> responding.
>>
>>
>> cheers,
>> stanislav
>>
>> On Thu, Feb 11, 2016 at 5:53 PM, Emrah <lists at kavun.ch> wrote:
>> > Hi list,
>> > I?m writing to gather your thoughts and suggestions on how to have a
>> high availability FS setup on different networks.
>> >
>> > I am trying to achieve the following:
>> >  - Load balance FreeSWITCH instances on 2 or more servers, possibly in
>> different countries.
>> > - Shared user directory and dialplan, but I?m not sure if shared
>> registrations would make sense.
>> > - If a server goes down, the phone should register on the alternative
>> servers. Obviously we can?t keep calls up.
>> >
>> > I?m obviously not the first one out there doing this. I?m trying to
>> learn from those who?ve come up with reliable solutions.
>> >
>> > I?ve tried sharing a registration table among multiple FS instances.
>> But it was a beginners mistake. Even with the right path to reach the
>> client, only the invites sent from the server used by the phone would be
>> processed.
>> > If my phone registers on server A, then server A shares the info with
>> server B, server B knows how to contact the phone but it won?t be able to.
>> Supposedly because of NAT issues.
>> >
>> > I am aiming for fully independent FS instances that can back each other
>> up and be used independently. I am guessing this would require some sort of
>> SBC or external registrar server with a Kamailio or Repro.
>> >
>> > Anyway just trying to spark the conversation around this subject and
>> hopefully we can come up with a formula that can help many with their FS
>> deployments. My provider?s network just went all down in IPv4 and HA behind
>> the same provider proved to be useless.
>> >
>> > Best,
>> > E
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 vagarwal at vertical.com  Fri Feb 12 01:00:57 2016
From: vagarwal at vertical.com (Varsha Agarwal)
Date: Thu, 11 Feb 2016 22:00:57 +0000
Subject: [Freeswitch-users] Freeswitch in high available clustered
	environment
Message-ID: <04AB0A185A23864CA9255FD38901F65D023374576B@SCEX1.vertical.com>

Hi All,

Is there a good documentation on how to setup Freeswitch in a clustered environment with a redundant node as well? I am looking through Wiki but there no one good article I found that has it all.

Thanks,
Varsha
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From steveayre at gmail.com  Fri Feb 12 01:41:33 2016
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 11 Feb 2016 22:41:33 +0000
Subject: [Freeswitch-users] Two Users Registered ..But calls only going
 one way
In-Reply-To: <CAK9Ws2dOCsLXVS=JPzj4yW-RV_SffNYrWYpoMn=AQy8x_x5gKw@mail.gmail.com>
References: <CAK9Ws2dOCsLXVS=JPzj4yW-RV_SffNYrWYpoMn=AQy8x_x5gKw@mail.gmail.com>
Message-ID: <CAFiqYukz1u+mQCKxiR8_ngYFtM2MM7C5VsD6Zz0dUpy-AOSFsg@mail.gmail.com>

See the contents of the Contact header in A's REGISTER messages. They tell
FreeSWITCH where to send the call to.

NAT can confuse matters when picking the correct value though. It needs to
be the external IP & port of the NAT router that the internal IP/port of
the SIP messages are mapped to. Sometimes the phone will put the internal
details instead which aren't routable externally. Sometimes it can detect
it correctly (eg via STUN). If it can't some routers will contain a SIP ALG
that will rewrite the header for a phone sending the internal ip/port to
the external ip/port, but sometimes this can cause more problems than it
solves if it doesn't do this correctly and it can't modify the packet if
you're using TLS.

On top of that that internal to external port mapping will expire on that
NAT router if you don't re-REGISTER frequently enough so that could stop
the INVITE getting through even if you're sending to the correct place.

If you're having issues like that getting the SIP packets through then it's
likely

If you can't fix it on the phone/router then you can also look at the NDLB
(no device left behind) options. For example there's one that'll use the
address the REGISTER is received from instead of the Contact header. This
differs from how SIP is supposed to work but works in most cases (usually a
phone will ask you to call it directly not via a proxy or route you
elsewhere).


On 11 February 2016 at 09:46, David Wafula <davidwaf at gmail.com> wrote:

> Hi all,
> I have two users who registered in the same domain: user A and B.
> A can call B just fine. When B tries to call A, there is silence (no
> ringback)..then after sometime the call goes into voice mail. A never
> receives the call.  Please see the call trace here:
>
>
> http://pastebin.com/gWrrS4zw
>
> Am not sure what is causing it.
>
> Regards
>
> --
> David W
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 blake at cogents.io  Fri Feb 12 01:45:34 2016
From: blake at cogents.io (Blake Priddy)
Date: Thu, 11 Feb 2016 16:45:34 -0600
Subject: [Freeswitch-users] Dev meeting
Message-ID: <CA+Kn03kAma9ENohtoWiLAwMMAdBTSgsRbuB7_C9UiHGA7kqiJw@mail.gmail.com>

https://www.gofundme.com/freeswitch

It's not too late!!
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From luis.daniel.lucio at gmail.com  Fri Feb 12 04:59:20 2016
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Thu, 11 Feb 2016 20:59:20 -0500
Subject: [Freeswitch-users] Freeswitch in high available clustered
	environment
In-Reply-To: <04AB0A185A23864CA9255FD38901F65D023374576B@SCEX1.vertical.com>
References: <04AB0A185A23864CA9255FD38901F65D023374576B@SCEX1.vertical.com>
Message-ID: <CAFLo2Qz3VenhqC6YJix42-6RZp9ARj3MqwYy=da0AK4WKssSJg@mail.gmail.com>

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

Enjoy, contact me offline of you need more
Le 11 f?vr. 2016 5:01 PM, "Varsha Agarwal" <vagarwal at vertical.com> a ?crit :

> Hi All,
>
>
>
> Is there a good documentation on how to setup Freeswitch in a clustered
> environment with a redundant node as well? I am looking through Wiki but
> there no one good article I found that has it all.
>
>
>
> Thanks,
>
> Varsha
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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.daniel.lucio at gmail.com  Fri Feb 12 05:40:09 2016
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Thu, 11 Feb 2016 21:40:09 -0500
Subject: [Freeswitch-users] High availability on different networks
In-Reply-To: <CAHtxdDdp=czW4sHONLs8mGpzYgLuNg1K4GSp2uKF_LA9_FgsUw@mail.gmail.com>
References: <78429801-FCA4-4BA9-8EA6-64A1BD7ECA28@kavun.ch>
	<CACOYK=gUYrGPouHq4voAygR7ctBOc6sdwBrAF8DWEeW6XkXS9A@mail.gmail.com>
	<CAC-LwPOdcwKvUYiR=1RzmrfdS9c_HHHnRg9QhFpsHU18coBjvw@mail.gmail.com>
	<CAHtxdDdp=czW4sHONLs8mGpzYgLuNg1K4GSp2uKF_LA9_FgsUw@mail.gmail.com>
Message-ID: <CAFLo2QxgPQE=BCV_8BdoZOXT507mgD=ZtFir-Y3HovguNuFduA@mail.gmail.com>

Read this
https://okay.com.mx/en/entrepreneurs/balancing-clustering-and-high-availability-with-fusionpbx.html

It is just what you're looking for. Take the ideas and modify it to your
needs
Le 11 f?vr. 2016 3:17 PM, "Sergey Safarov" <s.safarov at gmail.com> a ?crit :

> I use A record updated to point active FS servers in geo-distributed
> cluster.
> I DNS server i can recommend use Amazon Route53
>
> On Thu, Feb 11, 2016 at 10:27 PM, Vik Killa <vipkilla at gmail.com> wrote:
>
>> What about DNS SRV?
>>
>> On Thu, Feb 11, 2016 at 2:00 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
>> wrote:
>>
>>> hi Emrah and all,
>>>
>>> it's the first time I actually searched for it, but there are hosting
>>> offers with anycast IP routing. It means, you have multiple servers in
>>> various locations, and they share the same service IP address. The
>>> clients connect to the nearest server, which is determined by standard
>>> BGP routing. You are still limited to a single global hosting
>>> provider, but you benefit from its redundant network and geographical
>>> distribution.
>>>
>>> In case of anycast addressing, incoming connections will be served
>>> easily. But the outgoing connections are rather tricky: you will need
>>> to bring the outbound call to the physical server where the user has
>>> registered, and initiate the connection from its anycast address. So,
>>> you can share and replicate the registration database, but you need to
>>> send the outbound call to the server which accepted the registration.
>>> I guess you should be able to retrieve this information from the
>>> registration database. This needs to be looked in details.
>>>
>>> Google for anycast server hosting, and there are at least 3 providers
>>> offering virtual hosts, and OVH is offering physical hosts as well. I
>>> guess there are more providers with similar offerings.
>>>
>>>
>>> Without anycast, you would need to use redundant registrars sharing
>>> the same service IP address -- for example, Digitalocean offers such
>>> service within any single datacenter.
>>>
>>> Having multiple registrars with different IP addresses is also
>>> possible, but then you depend on the way how each particular SIP
>>> client handles multiple IP addresses after resolving the domain name.
>>> Some of them may get stuck to a single address, even if it's not
>>> responding.
>>>
>>>
>>> cheers,
>>> stanislav
>>>
>>> On Thu, Feb 11, 2016 at 5:53 PM, Emrah <lists at kavun.ch> wrote:
>>> > Hi list,
>>> > I?m writing to gather your thoughts and suggestions on how to have a
>>> high availability FS setup on different networks.
>>> >
>>> > I am trying to achieve the following:
>>> >  - Load balance FreeSWITCH instances on 2 or more servers, possibly in
>>> different countries.
>>> > - Shared user directory and dialplan, but I?m not sure if shared
>>> registrations would make sense.
>>> > - If a server goes down, the phone should register on the alternative
>>> servers. Obviously we can?t keep calls up.
>>> >
>>> > I?m obviously not the first one out there doing this. I?m trying to
>>> learn from those who?ve come up with reliable solutions.
>>> >
>>> > I?ve tried sharing a registration table among multiple FS instances.
>>> But it was a beginners mistake. Even with the right path to reach the
>>> client, only the invites sent from the server used by the phone would be
>>> processed.
>>> > If my phone registers on server A, then server A shares the info with
>>> server B, server B knows how to contact the phone but it won?t be able to.
>>> Supposedly because of NAT issues.
>>> >
>>> > I am aiming for fully independent FS instances that can back each
>>> other up and be used independently. I am guessing this would require some
>>> sort of SBC or external registrar server with a Kamailio or Repro.
>>> >
>>> > Anyway just trying to spark the conversation around this subject and
>>> hopefully we can come up with a formula that can help many with their FS
>>> deployments. My provider?s network just went all down in IPv4 and HA behind
>>> the same provider proved to be useless.
>>> >
>>> > Best,
>>> > E
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://confluence.freeswitch.org
>>> > http://www.cluecon.com
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 vagarwal at vertical.com  Fri Feb 12 05:59:25 2016
From: vagarwal at vertical.com (Varsha Agarwal)
Date: Fri, 12 Feb 2016 02:59:25 +0000
Subject: [Freeswitch-users] Freeswitch in high available
	clustered	environment
In-Reply-To: <CAFLo2Qz3VenhqC6YJix42-6RZp9ARj3MqwYy=da0AK4WKssSJg@mail.gmail.com>
References: <04AB0A185A23864CA9255FD38901F65D023374576B@SCEX1.vertical.com>,
	<CAFLo2Qz3VenhqC6YJix42-6RZp9ARj3MqwYy=da0AK4WKssSJg@mail.gmail.com>
Message-ID: <E7980948-763C-48C7-A0A7-CBEF9EBDD409@vertical.com>

Thanks I will review it.



On Feb 11, 2016, at 6:01 PM, Luis Daniel Lucio Quiroz <luis.daniel.lucio at gmail.com<mailto:luis.daniel.lucio at gmail.com>> wrote:


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

Enjoy, contact me offline of you need more

Le 11 f?vr. 2016 5:01 PM, "Varsha Agarwal" <vagarwal at vertical.com<mailto:vagarwal at vertical.com>> a ?crit :
Hi All,

Is there a good documentation on how to setup Freeswitch in a clustered environment with a redundant node as well? I am looking through Wiki but there no one good article I found that has it all.

Thanks,
Varsha

_________________________________________________________________________
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://lists.freeswitch.org/mailman/options/freeswitch-users>
http://www.freeswitch.org
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From max at nysolutions.com  Fri Feb 12 06:13:54 2016
From: max at nysolutions.com (Moishe Grunstein)
Date: Fri, 12 Feb 2016 03:13:54 +0000
Subject: [Freeswitch-users] High availability on different networks
In-Reply-To: <CAFLo2QxgPQE=BCV_8BdoZOXT507mgD=ZtFir-Y3HovguNuFduA@mail.gmail.com>
References: <78429801-FCA4-4BA9-8EA6-64A1BD7ECA28@kavun.ch>
	<CACOYK=gUYrGPouHq4voAygR7ctBOc6sdwBrAF8DWEeW6XkXS9A@mail.gmail.com>
	<CAC-LwPOdcwKvUYiR=1RzmrfdS9c_HHHnRg9QhFpsHU18coBjvw@mail.gmail.com>
	<CAHtxdDdp=czW4sHONLs8mGpzYgLuNg1K4GSp2uKF_LA9_FgsUw@mail.gmail.com>
	<CAFLo2QxgPQE=BCV_8BdoZOXT507mgD=ZtFir-Y3HovguNuFduA@mail.gmail.com>
Message-ID: <db5d3180005349a7ad8380bf93e2ecad@nysolutions.com>

Luis,
This mailing list is a Freeswitch users list, this list is for users to help each other with freeswitch, not a place to advertise your services, you may want to look at the freeswitch biz list http://lists.freeswitch.org/mailman/listinfo/freeswitch-biz


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 Luis Daniel Lucio Quiroz
Sent: Thursday, February 11, 2016 9:40 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] High availability on different networks


Read this
https://okay.com.mx/en/entrepreneurs/balancing-clustering-and-high-availability-with-fusionpbx.html

It is just what you're looking for. Take the ideas and modify it to your needs
Le 11 f?vr. 2016 3:17 PM, "Sergey Safarov" <s.safarov at gmail.com<mailto:s.safarov at gmail.com>> a ?crit :
I use A record updated to point active FS servers in geo-distributed cluster.
I DNS server i can recommend use Amazon Route53

On Thu, Feb 11, 2016 at 10:27 PM, Vik Killa <vipkilla at gmail.com<mailto:vipkilla at gmail.com>> wrote:
What about DNS SRV?

On Thu, Feb 11, 2016 at 2:00 PM, Stanislav Sinyagin <ssinyagin at gmail.com<mailto:ssinyagin at gmail.com>> wrote:
hi Emrah and all,

it's the first time I actually searched for it, but there are hosting
offers with anycast IP routing. It means, you have multiple servers in
various locations, and they share the same service IP address. The
clients connect to the nearest server, which is determined by standard
BGP routing. You are still limited to a single global hosting
provider, but you benefit from its redundant network and geographical
distribution.

In case of anycast addressing, incoming connections will be served
easily. But the outgoing connections are rather tricky: you will need
to bring the outbound call to the physical server where the user has
registered, and initiate the connection from its anycast address. So,
you can share and replicate the registration database, but you need to
send the outbound call to the server which accepted the registration.
I guess you should be able to retrieve this information from the
registration database. This needs to be looked in details.

Google for anycast server hosting, and there are at least 3 providers
offering virtual hosts, and OVH is offering physical hosts as well. I
guess there are more providers with similar offerings.


Without anycast, you would need to use redundant registrars sharing
the same service IP address -- for example, Digitalocean offers such
service within any single datacenter.

Having multiple registrars with different IP addresses is also
possible, but then you depend on the way how each particular SIP
client handles multiple IP addresses after resolving the domain name.
Some of them may get stuck to a single address, even if it's not
responding.


cheers,
stanislav

On Thu, Feb 11, 2016 at 5:53 PM, Emrah <lists at kavun.ch<mailto:lists at kavun.ch>> wrote:
> Hi list,
> I?m writing to gather your thoughts and suggestions on how to have a high availability FS setup on different networks.
>
> I am trying to achieve the following:
>  - Load balance FreeSWITCH instances on 2 or more servers, possibly in different countries.
> - Shared user directory and dialplan, but I?m not sure if shared registrations would make sense.
> - If a server goes down, the phone should register on the alternative servers. Obviously we can?t keep calls up.
>
> I?m obviously not the first one out there doing this. I?m trying to learn from those who?ve come up with reliable solutions.
>
> I?ve tried sharing a registration table among multiple FS instances. But it was a beginners mistake. Even with the right path to reach the client, only the invites sent from the server used by the phone would be processed.
> If my phone registers on server A, then server A shares the info with server B, server B knows how to contact the phone but it won?t be able to. Supposedly because of NAT issues.
>
> I am aiming for fully independent FS instances that can back each other up and be used independently. I am guessing this would require some sort of SBC or external registrar server with a Kamailio or Repro.
>
> Anyway just trying to spark the conversation around this subject and hopefully we can come up with a formula that can help many with their FS deployments. My provider?s network just went all down in IPv4 and HA behind the same provider proved to be useless.
>
> Best,
> E
> _________________________________________________________________________
> 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<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 bilaln018 at gmail.com  Fri Feb 12 09:06:38 2016
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Fri, 12 Feb 2016 11:06:38 +0500
Subject: [Freeswitch-users] [SDP Negotiation Error][Calling through
	JsSIP]
In-Reply-To: <CADSVerAwUUMTpzAJLXc0_6YMyYS1PzusQHeyHe6tFLzoucWyXg@mail.gmail.com>
References: <CADSVerCN05+5vvwvXKY=SrPhPwg8TKAWMUmXBv3P_y=tqmEr6A@mail.gmail.com>
	<CAEJMVkCrUZwSUhjM6s4+H5N28=i_0Qj+FoqjuAiDzYW0D1FCUA@mail.gmail.com>
	<CADSVerAwUUMTpzAJLXc0_6YMyYS1PzusQHeyHe6tFLzoucWyXg@mail.gmail.com>
Message-ID: <CADSVerBouUbj4+ZqGVjR8Fzx18JKZ90iVkd+mbjbDS7LaKRSgw@mail.gmail.com>

Hi All,
For the community archive i have added following in sofia internal  profile
<param name="apply-candidate-acl" value="rfc1918.auto"/>
and its started working.

Regards
Abbasi


On Thu, Feb 11, 2016 at 7:54 PM, Bilal Abbasi <bilaln018 at gmail.com> wrote:

> Thanks brain for reply but what i need to do, how can i get rid of this
> warning.
>
> many thanks
>
> abbasi
>
>
> On Thursday, February 11, 2016, Brian West <brian at freeswitch.org> wrote:
>
> NO candidate ACL defined, Defaulting to wan.auto
>> sofia/internal/1001 at 192.241.213.201:7000 no suitable candidates found.
>>
>> On Thu, Feb 11, 2016 at 3:19 AM, Bilal Abbasi <bilaln018 at gmail.com>
>> wrote:
>>
>>> Hi all,
>>>
>>> Currently i am facing an CODEC NEGOTIATION ERROR while calling through
>>> JsSIP Caller 1001 Callie 1000,
>>> Please view the logs,
>>> https://pastebin.freeswitch.org/24550
>>>
>>> I have enabled PCMA and PCMU in my var.xml,
>>>
>>> Regards
>>> Abbasi
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> 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 ssinyagin at gmail.com  Fri Feb 12 10:40:53 2016
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Fri, 12 Feb 2016 08:40:53 +0100
Subject: [Freeswitch-users] High availability on different networks
In-Reply-To: <CACOYK=gUYrGPouHq4voAygR7ctBOc6sdwBrAF8DWEeW6XkXS9A@mail.gmail.com>
References: <78429801-FCA4-4BA9-8EA6-64A1BD7ECA28@kavun.ch>
	<CACOYK=gUYrGPouHq4voAygR7ctBOc6sdwBrAF8DWEeW6XkXS9A@mail.gmail.com>
Message-ID: <CACOYK=jtfv356yOWXSJuqcVNL0L+_rMWAiQKEWqAC9i8bOwqUg@mail.gmail.com>

there is an issue with anycast routing though: when you bring up a new
server, some running sessions will be dropped, because their IP
packets would be routed to a different host. So, it needs a careful
design. Maybe place only the SIP proxy on an anycast address, and run
the calls from unique local addresses.


Multiple DNS SRV records with different priorities are also possible,
but you can't direct the users to the nearest location within the same
domain. Also a bunch of SIP clients needs to be tested and you need to
answer the questions, like:

-- what is the timeout if the primary server is unavailable?
-- if the primary host goes down during the call, how soon can the
client re-dial?
-- what happens if the primary server comes up again?






On Thu, Feb 11, 2016 at 8:00 PM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
> hi Emrah and all,
>
> it's the first time I actually searched for it, but there are hosting
> offers with anycast IP routing. It means, you have multiple servers in
> various locations, and they share the same service IP address. The
> clients connect to the nearest server, which is determined by standard
> BGP routing. You are still limited to a single global hosting
> provider, but you benefit from its redundant network and geographical
> distribution.
>
> In case of anycast addressing, incoming connections will be served
> easily. But the outgoing connections are rather tricky: you will need
> to bring the outbound call to the physical server where the user has
> registered, and initiate the connection from its anycast address. So,
> you can share and replicate the registration database, but you need to
> send the outbound call to the server which accepted the registration.
> I guess you should be able to retrieve this information from the
> registration database. This needs to be looked in details.
>
> Google for anycast server hosting, and there are at least 3 providers
> offering virtual hosts, and OVH is offering physical hosts as well. I
> guess there are more providers with similar offerings.
>
>
> Without anycast, you would need to use redundant registrars sharing
> the same service IP address -- for example, Digitalocean offers such
> service within any single datacenter.
>
> Having multiple registrars with different IP addresses is also
> possible, but then you depend on the way how each particular SIP
> client handles multiple IP addresses after resolving the domain name.
> Some of them may get stuck to a single address, even if it's not
> responding.
>
>
> cheers,
> stanislav
>
> On Thu, Feb 11, 2016 at 5:53 PM, Emrah <lists at kavun.ch> wrote:
>> Hi list,
>> I?m writing to gather your thoughts and suggestions on how to have a high availability FS setup on different networks.
>>
>> I am trying to achieve the following:
>>  - Load balance FreeSWITCH instances on 2 or more servers, possibly in different countries.
>> - Shared user directory and dialplan, but I?m not sure if shared registrations would make sense.
>> - If a server goes down, the phone should register on the alternative servers. Obviously we can?t keep calls up.
>>
>> I?m obviously not the first one out there doing this. I?m trying to learn from those who?ve come up with reliable solutions.
>>
>> I?ve tried sharing a registration table among multiple FS instances. But it was a beginners mistake. Even with the right path to reach the client, only the invites sent from the server used by the phone would be processed.
>> If my phone registers on server A, then server A shares the info with server B, server B knows how to contact the phone but it won?t be able to. Supposedly because of NAT issues.
>>
>> I am aiming for fully independent FS instances that can back each other up and be used independently. I am guessing this would require some sort of SBC or external registrar server with a Kamailio or Repro.
>>
>> Anyway just trying to spark the conversation around this subject and hopefully we can come up with a formula that can help many with their FS deployments. My provider?s network just went all down in IPv4 and HA behind the same provider proved to be useless.
>>
>> Best,
>> E
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 andrew at cassidywebservices.co.uk  Fri Feb 12 12:09:32 2016
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Fri, 12 Feb 2016 09:09:32 +0000
Subject: [Freeswitch-users] High availability on different networks
In-Reply-To: <CACOYK=jtfv356yOWXSJuqcVNL0L+_rMWAiQKEWqAC9i8bOwqUg@mail.gmail.com>
References: <78429801-FCA4-4BA9-8EA6-64A1BD7ECA28@kavun.ch>
	<CACOYK=gUYrGPouHq4voAygR7ctBOc6sdwBrAF8DWEeW6XkXS9A@mail.gmail.com>
	<CACOYK=jtfv356yOWXSJuqcVNL0L+_rMWAiQKEWqAC9i8bOwqUg@mail.gmail.com>
Message-ID: <CAMbW9oJxHhWjcVvV7TdtzHMaxWGsu4L8kG-8YYmaNKXJPObgNQ@mail.gmail.com>

It's not instant, but I've used OVH failover IP's to do that sort of thing.
Heartbeat over ipsec tunnel, use pacemaker with a custom script that does
the OVH API call to move the IP address.

Sadly it's not that quick, takes about 2 minutes.

On 12 February 2016 at 07:40, Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> there is an issue with anycast routing though: when you bring up a new
> server, some running sessions will be dropped, because their IP
> packets would be routed to a different host. So, it needs a careful
> design. Maybe place only the SIP proxy on an anycast address, and run
> the calls from unique local addresses.
>
>
> Multiple DNS SRV records with different priorities are also possible,
> but you can't direct the users to the nearest location within the same
> domain. Also a bunch of SIP clients needs to be tested and you need to
> answer the questions, like:
>
> -- what is the timeout if the primary server is unavailable?
> -- if the primary host goes down during the call, how soon can the
> client re-dial?
> -- what happens if the primary server comes up again?
>
>
>
>
>
>
> On Thu, Feb 11, 2016 at 8:00 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
> > hi Emrah and all,
> >
> > it's the first time I actually searched for it, but there are hosting
> > offers with anycast IP routing. It means, you have multiple servers in
> > various locations, and they share the same service IP address. The
> > clients connect to the nearest server, which is determined by standard
> > BGP routing. You are still limited to a single global hosting
> > provider, but you benefit from its redundant network and geographical
> > distribution.
> >
> > In case of anycast addressing, incoming connections will be served
> > easily. But the outgoing connections are rather tricky: you will need
> > to bring the outbound call to the physical server where the user has
> > registered, and initiate the connection from its anycast address. So,
> > you can share and replicate the registration database, but you need to
> > send the outbound call to the server which accepted the registration.
> > I guess you should be able to retrieve this information from the
> > registration database. This needs to be looked in details.
> >
> > Google for anycast server hosting, and there are at least 3 providers
> > offering virtual hosts, and OVH is offering physical hosts as well. I
> > guess there are more providers with similar offerings.
> >
> >
> > Without anycast, you would need to use redundant registrars sharing
> > the same service IP address -- for example, Digitalocean offers such
> > service within any single datacenter.
> >
> > Having multiple registrars with different IP addresses is also
> > possible, but then you depend on the way how each particular SIP
> > client handles multiple IP addresses after resolving the domain name.
> > Some of them may get stuck to a single address, even if it's not
> > responding.
> >
> >
> > cheers,
> > stanislav
> >
> > On Thu, Feb 11, 2016 at 5:53 PM, Emrah <lists at kavun.ch> wrote:
> >> Hi list,
> >> I?m writing to gather your thoughts and suggestions on how to have a
> high availability FS setup on different networks.
> >>
> >> I am trying to achieve the following:
> >>  - Load balance FreeSWITCH instances on 2 or more servers, possibly in
> different countries.
> >> - Shared user directory and dialplan, but I?m not sure if shared
> registrations would make sense.
> >> - If a server goes down, the phone should register on the alternative
> servers. Obviously we can?t keep calls up.
> >>
> >> I?m obviously not the first one out there doing this. I?m trying to
> learn from those who?ve come up with reliable solutions.
> >>
> >> I?ve tried sharing a registration table among multiple FS instances.
> But it was a beginners mistake. Even with the right path to reach the
> client, only the invites sent from the server used by the phone would be
> processed.
> >> If my phone registers on server A, then server A shares the info with
> server B, server B knows how to contact the phone but it won?t be able to.
> Supposedly because of NAT issues.
> >>
> >> I am aiming for fully independent FS instances that can back each other
> up and be used independently. I am guessing this would require some sort of
> SBC or external registrar server with a Kamailio or Repro.
> >>
> >> Anyway just trying to spark the conversation around this subject and
> hopefully we can come up with a formula that can help many with their FS
> deployments. My provider?s network just went all down in IPv4 and HA behind
> the same provider proved to be useless.
> >>
> >> Best,
> >> E
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960  *F
<info at cassidywebservices.co.uk> *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From aqsyounas at gmail.com  Fri Feb 12 12:14:36 2016
From: aqsyounas at gmail.com (Aqs Younas)
Date: Fri, 12 Feb 2016 14:14:36 +0500
Subject: [Freeswitch-users] How to add both rpid and pid in freeswitch.
Message-ID: <CAGVOjVWtRwEGHf3Mk-6YE-u_VEpm=bGVXE3Sbnjau+fOqWHN-Q@mail.gmail.com>

Hi,

I am bridging Invite containing both RPID and PID to some destination. Like

 <action application="bridge"
data="{sip_cid_type=rpid,origination_caller_id_name=_undef_}sofia/external/${sip_req_uri};${sip_req_params};fs_path=sip:${Dest}"/>

Currently, I see freeswitch can add only one header PID or RPID but not
both with  sip_cid_type variable.

How can i Add both?

Remote-Party-ID: <sip:14248271614 at 192.168.10.25>;privacy=off;screen=yes
P-Asserted-Identity: <sip:14248271614 at 192.168.10.25;user=phone>

Any suggestion is much appreciate.

Best Regards.
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From aqsyounas at gmail.com  Fri Feb 12 13:50:48 2016
From: aqsyounas at gmail.com (Aqs Younas)
Date: Fri, 12 Feb 2016 15:50:48 +0500
Subject: [Freeswitch-users] fs_path adding route header
Message-ID: <CAGVOjVWKQ9Cebg=_5eMYXd4FCM21ZHsATW61XHqn29Qjr1B5uA@mail.gmail.com>

Hi,

I am using fs_path to proxy Invite to specified proxy. But I see fs_path
adding route header in Invite. As B2BUA freeswitch must be generating new
call instead of adding route header.

Is this intended behavior? Or something wrong with my configuration

Best Regards.
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From ssinyagin at gmail.com  Fri Feb 12 14:49:17 2016
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Fri, 12 Feb 2016 12:49:17 +0100
Subject: [Freeswitch-users] High availability on different networks
In-Reply-To: <CAMbW9oJxHhWjcVvV7TdtzHMaxWGsu4L8kG-8YYmaNKXJPObgNQ@mail.gmail.com>
References: <78429801-FCA4-4BA9-8EA6-64A1BD7ECA28@kavun.ch>
	<CACOYK=gUYrGPouHq4voAygR7ctBOc6sdwBrAF8DWEeW6XkXS9A@mail.gmail.com>
	<CACOYK=jtfv356yOWXSJuqcVNL0L+_rMWAiQKEWqAC9i8bOwqUg@mail.gmail.com>
	<CAMbW9oJxHhWjcVvV7TdtzHMaxWGsu4L8kG-8YYmaNKXJPObgNQ@mail.gmail.com>
Message-ID: <CACOYK=iefwzaSjdN0Xk2PCCh74_naSq5Kqu1WZ7Ae03Ena9h7w@mail.gmail.com>

as Sergey has proposed, there could be a DNS service which monitors the
availability of your VoIP servers and changes the DNS entries if a server
goes down. The TTL for individual SRV records could be set to few seconds.

But that means again that all users are using the same server, so it's not
really a distributed model as Emrah challenged in the original mail.

Probably this new project will help in building a distributed cluster, but
it needs a detailed study:  https://ipfs.io/






On Fri, Feb 12, 2016 at 10:09 AM, Andrew Cassidy <
andrew at cassidywebservices.co.uk> wrote:

> It's not instant, but I've used OVH failover IP's to do that sort of
> thing. Heartbeat over ipsec tunnel, use pacemaker with a custom script that
> does the OVH API call to move the IP address.
>
> Sadly it's not that quick, takes about 2 minutes.
>
> On 12 February 2016 at 07:40, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
>
>> there is an issue with anycast routing though: when you bring up a new
>> server, some running sessions will be dropped, because their IP
>> packets would be routed to a different host. So, it needs a careful
>> design. Maybe place only the SIP proxy on an anycast address, and run
>> the calls from unique local addresses.
>>
>>
>> Multiple DNS SRV records with different priorities are also possible,
>> but you can't direct the users to the nearest location within the same
>> domain. Also a bunch of SIP clients needs to be tested and you need to
>> answer the questions, like:
>>
>> -- what is the timeout if the primary server is unavailable?
>> -- if the primary host goes down during the call, how soon can the
>> client re-dial?
>> -- what happens if the primary server comes up again?
>>
>>
>>
>>
>>
>>
>> On Thu, Feb 11, 2016 at 8:00 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
>> wrote:
>> > hi Emrah and all,
>> >
>> > it's the first time I actually searched for it, but there are hosting
>> > offers with anycast IP routing. It means, you have multiple servers in
>> > various locations, and they share the same service IP address. The
>> > clients connect to the nearest server, which is determined by standard
>> > BGP routing. You are still limited to a single global hosting
>> > provider, but you benefit from its redundant network and geographical
>> > distribution.
>> >
>> > In case of anycast addressing, incoming connections will be served
>> > easily. But the outgoing connections are rather tricky: you will need
>> > to bring the outbound call to the physical server where the user has
>> > registered, and initiate the connection from its anycast address. So,
>> > you can share and replicate the registration database, but you need to
>> > send the outbound call to the server which accepted the registration.
>> > I guess you should be able to retrieve this information from the
>> > registration database. This needs to be looked in details.
>> >
>> > Google for anycast server hosting, and there are at least 3 providers
>> > offering virtual hosts, and OVH is offering physical hosts as well. I
>> > guess there are more providers with similar offerings.
>> >
>> >
>> > Without anycast, you would need to use redundant registrars sharing
>> > the same service IP address -- for example, Digitalocean offers such
>> > service within any single datacenter.
>> >
>> > Having multiple registrars with different IP addresses is also
>> > possible, but then you depend on the way how each particular SIP
>> > client handles multiple IP addresses after resolving the domain name.
>> > Some of them may get stuck to a single address, even if it's not
>> > responding.
>> >
>> >
>> > cheers,
>> > stanislav
>> >
>> > On Thu, Feb 11, 2016 at 5:53 PM, Emrah <lists at kavun.ch> wrote:
>> >> Hi list,
>> >> I?m writing to gather your thoughts and suggestions on how to have a
>> high availability FS setup on different networks.
>> >>
>> >> I am trying to achieve the following:
>> >>  - Load balance FreeSWITCH instances on 2 or more servers, possibly in
>> different countries.
>> >> - Shared user directory and dialplan, but I?m not sure if shared
>> registrations would make sense.
>> >> - If a server goes down, the phone should register on the alternative
>> servers. Obviously we can?t keep calls up.
>> >>
>> >> I?m obviously not the first one out there doing this. I?m trying to
>> learn from those who?ve come up with reliable solutions.
>> >>
>> >> I?ve tried sharing a registration table among multiple FS instances.
>> But it was a beginners mistake. Even with the right path to reach the
>> client, only the invites sent from the server used by the phone would be
>> processed.
>> >> If my phone registers on server A, then server A shares the info with
>> server B, server B knows how to contact the phone but it won?t be able to.
>> Supposedly because of NAT issues.
>> >>
>> >> I am aiming for fully independent FS instances that can back each
>> other up and be used independently. I am guessing this would require some
>> sort of SBC or external registrar server with a Kamailio or Repro.
>> >>
>> >> Anyway just trying to spark the conversation around this subject and
>> hopefully we can come up with a formula that can help many with their FS
>> deployments. My provider?s network just went all down in IPv4 and HA behind
>> the same provider proved to be useless.
>> >>
>> >> Best,
>> >> E
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://confluence.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
>
>
> *T <info at cassidywebservices.co.uk> *03300 100 960  *F
> <info at cassidywebservices.co.uk> *03300 100 961
> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 blake at cogents.io  Fri Feb 12 16:44:44 2016
From: blake at cogents.io (Blake Priddy)
Date: Fri, 12 Feb 2016 07:44:44 -0600
Subject: [Freeswitch-users] Over the half way mark!!! :D
In-Reply-To: <CA+Kn03nHsys=VWogJLG-98cGoLeG=YmhXqF7xjucFt6uX8vw9g@mail.gmail.com>
References: <CA+Kn03nHsys=VWogJLG-98cGoLeG=YmhXqF7xjucFt6uX8vw9g@mail.gmail.com>
Message-ID: <CA+Kn03m3OW=w3qtZg=tUzDJztD81f79maXAuPcwreop36RZvmQ@mail.gmail.com>

They are over the half way mark! Thanks y'all. Keep it coming!! :)

https://www.gofundme.com/freeswitch
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From mike at jerris.com  Fri Feb 12 19:03:33 2016
From: mike at jerris.com (Michael Jerris)
Date: Fri, 12 Feb 2016 11:03:33 -0500
Subject: [Freeswitch-users] How to add both rpid and pid in freeswitch.
In-Reply-To: <CAGVOjVWtRwEGHf3Mk-6YE-u_VEpm=bGVXE3Sbnjau+fOqWHN-Q@mail.gmail.com>
References: <CAGVOjVWtRwEGHf3Mk-6YE-u_VEpm=bGVXE3Sbnjau+fOqWHN-Q@mail.gmail.com>
Message-ID: <CAEwTsAhyNSew+6XrWYF3Xs0o3rjzki+=JeXH_dMt=dzOJ7KMPg@mail.gmail.com>

I don't think we have any way exposed to do this right now, but I wouldn't
be opposed to a patch that added this

On Friday, February 12, 2016, Aqs Younas <aqsyounas at gmail.com> wrote:

> Hi,
>
> I am bridging Invite containing both RPID and PID to some destination. Like
>
>  <action application="bridge"
> data="{sip_cid_type=rpid,origination_caller_id_name=_undef_}sofia/external/${sip_req_uri};${sip_req_params};fs_path=sip:${Dest}"/>
>
> Currently, I see freeswitch can add only one header PID or RPID but not
> both with  sip_cid_type variable.
>
> How can i Add both?
>
> Remote-Party-ID: <sip:14248271614 at 192.168.10.25
> <javascript:_e(%7B%7D,'cvml','sip:14248271614 at 192.168.10.25');>
> >;privacy=off;screen=yes
> P-Asserted-Identity: <sip:14248271614 at 192.168.10.25
> <javascript:_e(%7B%7D,'cvml','sip:14248271614 at 192.168.10.25');>
> ;user=phone>
>
> Any suggestion is much appreciate.
>
> Best Regards.
>
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From s.safarov at gmail.com  Fri Feb 12 19:14:44 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 12 Feb 2016 16:14:44 +0000
Subject: [Freeswitch-users] High availability on different networks
In-Reply-To: <CACOYK=iefwzaSjdN0Xk2PCCh74_naSq5Kqu1WZ7Ae03Ena9h7w@mail.gmail.com>
References: <78429801-FCA4-4BA9-8EA6-64A1BD7ECA28@kavun.ch>
	<CACOYK=gUYrGPouHq4voAygR7ctBOc6sdwBrAF8DWEeW6XkXS9A@mail.gmail.com>
	<CACOYK=jtfv356yOWXSJuqcVNL0L+_rMWAiQKEWqAC9i8bOwqUg@mail.gmail.com>
	<CAMbW9oJxHhWjcVvV7TdtzHMaxWGsu4L8kG-8YYmaNKXJPObgNQ@mail.gmail.com>
	<CACOYK=iefwzaSjdN0Xk2PCCh74_naSq5Kqu1WZ7Ae03Ena9h7w@mail.gmail.com>
Message-ID: <CAHtxdDfysRf9kMNYcbrYyjF_bXdD1E=ffB=+bmV0A=ne85ByxA@mail.gmail.com>

As cluster solution I use kazoo cluster with kamailio - after 6 mouth of
usage I can say it perfect solution.

Also updates DNS record can be implemented via custom corosync resource
agent script.  Write this script is task of two or tree days. This solution
is not require external servers.

On Fri, Feb 12, 2016, 14:50 Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

> as Sergey has proposed, there could be a DNS service which monitors the
> availability of your VoIP servers and changes the DNS entries if a server
> goes down. The TTL for individual SRV records could be set to few seconds.
>
> But that means again that all users are using the same server, so it's not
> really a distributed model as Emrah challenged in the original mail.
>
> Probably this new project will help in building a distributed cluster, but
> it needs a detailed study:  https://ipfs.io/
>
>
>
>
>
>
> On Fri, Feb 12, 2016 at 10:09 AM, Andrew Cassidy <
> andrew at cassidywebservices.co.uk> wrote:
>
>> It's not instant, but I've used OVH failover IP's to do that sort of
>> thing. Heartbeat over ipsec tunnel, use pacemaker with a custom script that
>> does the OVH API call to move the IP address.
>>
>> Sadly it's not that quick, takes about 2 minutes.
>>
>> On 12 February 2016 at 07:40, Stanislav Sinyagin <ssinyagin at gmail.com>
>> wrote:
>>
>>> there is an issue with anycast routing though: when you bring up a new
>>> server, some running sessions will be dropped, because their IP
>>> packets would be routed to a different host. So, it needs a careful
>>> design. Maybe place only the SIP proxy on an anycast address, and run
>>> the calls from unique local addresses.
>>>
>>>
>>> Multiple DNS SRV records with different priorities are also possible,
>>> but you can't direct the users to the nearest location within the same
>>> domain. Also a bunch of SIP clients needs to be tested and you need to
>>> answer the questions, like:
>>>
>>> -- what is the timeout if the primary server is unavailable?
>>> -- if the primary host goes down during the call, how soon can the
>>> client re-dial?
>>> -- what happens if the primary server comes up again?
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Thu, Feb 11, 2016 at 8:00 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
>>> wrote:
>>> > hi Emrah and all,
>>> >
>>> > it's the first time I actually searched for it, but there are hosting
>>> > offers with anycast IP routing. It means, you have multiple servers in
>>> > various locations, and they share the same service IP address. The
>>> > clients connect to the nearest server, which is determined by standard
>>> > BGP routing. You are still limited to a single global hosting
>>> > provider, but you benefit from its redundant network and geographical
>>> > distribution.
>>> >
>>> > In case of anycast addressing, incoming connections will be served
>>> > easily. But the outgoing connections are rather tricky: you will need
>>> > to bring the outbound call to the physical server where the user has
>>> > registered, and initiate the connection from its anycast address. So,
>>> > you can share and replicate the registration database, but you need to
>>> > send the outbound call to the server which accepted the registration.
>>> > I guess you should be able to retrieve this information from the
>>> > registration database. This needs to be looked in details.
>>> >
>>> > Google for anycast server hosting, and there are at least 3 providers
>>> > offering virtual hosts, and OVH is offering physical hosts as well. I
>>> > guess there are more providers with similar offerings.
>>> >
>>> >
>>> > Without anycast, you would need to use redundant registrars sharing
>>> > the same service IP address -- for example, Digitalocean offers such
>>> > service within any single datacenter.
>>> >
>>> > Having multiple registrars with different IP addresses is also
>>> > possible, but then you depend on the way how each particular SIP
>>> > client handles multiple IP addresses after resolving the domain name.
>>> > Some of them may get stuck to a single address, even if it's not
>>> > responding.
>>> >
>>> >
>>> > cheers,
>>> > stanislav
>>> >
>>> > On Thu, Feb 11, 2016 at 5:53 PM, Emrah <lists at kavun.ch> wrote:
>>> >> Hi list,
>>> >> I?m writing to gather your thoughts and suggestions on how to have a
>>> high availability FS setup on different networks.
>>> >>
>>> >> I am trying to achieve the following:
>>> >>  - Load balance FreeSWITCH instances on 2 or more servers, possibly
>>> in different countries.
>>> >> - Shared user directory and dialplan, but I?m not sure if shared
>>> registrations would make sense.
>>> >> - If a server goes down, the phone should register on the alternative
>>> servers. Obviously we can?t keep calls up.
>>> >>
>>> >> I?m obviously not the first one out there doing this. I?m trying to
>>> learn from those who?ve come up with reliable solutions.
>>> >>
>>> >> I?ve tried sharing a registration table among multiple FS instances.
>>> But it was a beginners mistake. Even with the right path to reach the
>>> client, only the invites sent from the server used by the phone would be
>>> processed.
>>> >> If my phone registers on server A, then server A shares the info with
>>> server B, server B knows how to contact the phone but it won?t be able to.
>>> Supposedly because of NAT issues.
>>> >>
>>> >> I am aiming for fully independent FS instances that can back each
>>> other up and be used independently. I am guessing this would require some
>>> sort of SBC or external registrar server with a Kamailio or Repro.
>>> >>
>>> >> Anyway just trying to spark the conversation around this subject and
>>> hopefully we can come up with a formula that can help many with their FS
>>> deployments. My provider?s network just went all down in IPv4 and HA behind
>>> the same provider proved to be useless.
>>> >>
>>> >> Best,
>>> >> E
>>> >>
>>> _________________________________________________________________________
>>> >> Professional FreeSWITCH Consulting Services:
>>> >> consulting at freeswitch.org
>>> >> http://www.freeswitchsolutions.com
>>> >>
>>> >> Official FreeSWITCH Sites
>>> >> http://www.freeswitch.org
>>> >> http://confluence.freeswitch.org
>>> >> http://www.cluecon.com
>>> >>
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>> Managing Director
>>
>>
>> *T <info at cassidywebservices.co.uk> *03300 100 960  *F
>> <info at cassidywebservices.co.uk> *03300 100 961
>> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
>> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From gmaruzz at gmail.com  Fri Feb 12 19:25:00 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 12 Feb 2016 17:25:00 +0100
Subject: [Freeswitch-users] FUND THE PARTY! @ FreeSWITCH Annual Core Devs
	Meeting
Message-ID: <CALXCt0qm9hAQx5--nXzvm0nS3PuqpJVTXiLsFzi94Tt-5LxYig@mail.gmail.com>

How much we all made in 2015 thanks to FreeSWITCH?

How much we made thanks to core devs (Anthony, Mike, Ken, Brian, William,
etc) answering our difficult questions?

Shovel your HARD EARNED MONEY to our HARD PARTYING and HARD CODING CORE
DEVS !

BUY THEM LUNCH AND REFRESHMENTS !

HERE! ==> https://www.gofundme.com/freeswitch

If not now, when?

Because!

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From aqsyounas at gmail.com  Fri Feb 12 19:35:26 2016
From: aqsyounas at gmail.com (Aqs Younas)
Date: Fri, 12 Feb 2016 21:35:26 +0500
Subject: [Freeswitch-users] How to add both rpid and pid in freeswitch.
In-Reply-To: <CAEwTsAhyNSew+6XrWYF3Xs0o3rjzki+=JeXH_dMt=dzOJ7KMPg@mail.gmail.com>
References: <CAGVOjVWtRwEGHf3Mk-6YE-u_VEpm=bGVXE3Sbnjau+fOqWHN-Q@mail.gmail.com>
	<CAEwTsAhyNSew+6XrWYF3Xs0o3rjzki+=JeXH_dMt=dzOJ7KMPg@mail.gmail.com>
Message-ID: <CAGVOjVU-0L+y2A1MNf38iY2d3PVOW04wZkCKs0ju4hzwTtsr6A@mail.gmail.com>

Thanks Micheal.
I am able to achieve above by setting paid as header and adding rpid
as sip_cid_type
in bridge command.

On 12 February 2016 at 21:03, Michael Jerris <mike at jerris.com> wrote:

> I don't think we have any way exposed to do this right now, but I wouldn't
> be opposed to a patch that added this
>
>
> On Friday, February 12, 2016, Aqs Younas <aqsyounas at gmail.com> wrote:
>
>> Hi,
>>
>> I am bridging Invite containing both RPID and PID to some destination.
>> Like
>>
>>  <action application="bridge"
>> data="{sip_cid_type=rpid,origination_caller_id_name=_undef_}sofia/external/${sip_req_uri};${sip_req_params};fs_path=sip:${Dest}"/>
>>
>> Currently, I see freeswitch can add only one header PID or RPID but not
>> both with  sip_cid_type variable.
>>
>> How can i Add both?
>>
>> Remote-Party-ID: <sip:14248271614 at 192.168.10.25>;privacy=off;screen=yes
>> P-Asserted-Identity: <sip:14248271614 at 192.168.10.25;user=phone>
>>
>> Any suggestion is much appreciate.
>>
>> 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 aqsyounas at gmail.com  Fri Feb 12 21:24:26 2016
From: aqsyounas at gmail.com (Aqs Younas)
Date: Fri, 12 Feb 2016 23:24:26 +0500
Subject: [Freeswitch-users] fs_path adding route header
In-Reply-To: <CAGVOjVWKQ9Cebg=_5eMYXd4FCM21ZHsATW61XHqn29Qjr1B5uA@mail.gmail.com>
References: <CAGVOjVWKQ9Cebg=_5eMYXd4FCM21ZHsATW61XHqn29Qjr1B5uA@mail.gmail.com>
Message-ID: <CAGVOjVV8dap=fNt0SfqnF-5SF36sW+5K5LXMj0HxMDS6j4aTNg@mail.gmail.com>

freeswitch is adding route header of value fs_path. Which is causing
problem in 200ok and ack. Is there any way to proxy request without making
freeswitch add route header. Or any other way to make proxy request to
destination other than RURI without changing RURI.

Best Regards.

On 12 February 2016 at 15:50, Aqs Younas <aqsyounas at gmail.com> wrote:

> Hi,
>
> I am using fs_path to proxy Invite to specified proxy. But I see fs_path
> adding route header in Invite. As B2BUA freeswitch must be generating new
> call instead of adding route header.
>
> Is this intended behavior? Or something wrong with my configuration
>
> Best Regards.
>
>
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From lists at telefaks.de  Fri Feb 12 22:56:17 2016
From: lists at telefaks.de (Peter Steinbach)
Date: Fri, 12 Feb 2016 20:56:17 +0100
Subject: [Freeswitch-users] FUND THE PARTY! @ FreeSWITCH Annual Core
 Devs Meeting
In-Reply-To: <CALXCt0qm9hAQx5--nXzvm0nS3PuqpJVTXiLsFzi94Tt-5LxYig@mail.gmail.com>
References: <CALXCt0qm9hAQx5--nXzvm0nS3PuqpJVTXiLsFzi94Tt-5LxYig@mail.gmail.com>
Message-ID: <56BE38E1.6070303@telefaks.de>

Just transferred another thank you via Paypal.

Enjoy your meal!

Best regards
Peter


On 02/12/16 17:25, Giovanni Maruzzelli wrote:
> How much we all made in 2015 thanks to FreeSWITCH?
>
> How much we made thanks to core devs (Anthony, Mike, Ken, Brian,
> William, etc) answering our difficult questions?
>
> Shovel your HARD EARNED MONEY to our HARD PARTYING and HARD CODING
> CORE DEVS !
>
> BUY THEM LUNCH AND REFRESHMENTS !
>
> HERE! ==> https://www.gofundme.com/freeswitch
>
> If not now, when?
>
> Because!
>
> -- 
> 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


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de

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From brian at freeswitch.org  Sat Feb 13 02:50:18 2016
From: brian at freeswitch.org (Brian West)
Date: Fri, 12 Feb 2016 17:50:18 -0600
Subject: [Freeswitch-users] Freeswitch in high available clustered
	environment
In-Reply-To: <CAFLo2Qz3VenhqC6YJix42-6RZp9ARj3MqwYy=da0AK4WKssSJg@mail.gmail.com>
References: <04AB0A185A23864CA9255FD38901F65D023374576B@SCEX1.vertical.com>
	<CAFLo2Qz3VenhqC6YJix42-6RZp9ARj3MqwYy=da0AK4WKssSJg@mail.gmail.com>
Message-ID: <CAEJMVkCvACFRg6VVnHTkjx1k0t8_RSRiqY+pANz_Jg_gPf2m=w@mail.gmail.com>

We're actually working on a doc for this, it should be out soon.

On Thu, Feb 11, 2016 at 7:59 PM, Luis Daniel Lucio Quiroz <
luis.daniel.lucio at gmail.com> wrote:

> http://inside-out.xyz/technology/how-to-configure-freeswitch-for-ha.html
>
> Enjoy, contact me offline of you need more
> Le 11 f?vr. 2016 5:01 PM, "Varsha Agarwal" <vagarwal at vertical.com> a
> ?crit :
>
>> Hi All,
>>
>>
>>
>> Is there a good documentation on how to setup Freeswitch in a clustered
>> environment with a redundant node as well? I am looking through Wiki but
>> there no one good article I found that has it all.
>>
>>
>>
>> Thanks,
>>
>> Varsha
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

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 568691 at gmail.com  Sat Feb 13 04:11:54 2016
From: 568691 at gmail.com (Alexandru Covalschi)
Date: Sat, 13 Feb 2016 03:11:54 +0200
Subject: [Freeswitch-users] Get actual duration of the call after hangup
Message-ID: <CAD_iGr-M-X6ERmm7=foUPfc_jx6FTzu9Ayy+5YtMSPtW_qWD1w@mail.gmail.com>

Hello folks!

I have a problem - I need to get actual duration of the call - I mean from
ANSWER to HANGUP, without the time user waited being parked on the bridge.
The thing is I need to write that info into a custom table which belongs to
a custom database where I have leg_a uuid.
How can I achieve that in case of:
1. Outoing origination via bridge()
2. Connecting two customers which are already inside a freeswitch via
bridge()
I've read about mod_cdr_csv but it afaik doesn't allow to connect to a
custom db.
Would mod_odbc_cdr allow me to do that? And if yes - can you tell me
"chan-var-name"?

And don't be afraid - such slow method is not for billing :) Just to
provide customer more detailed calls history.

Thanks!
-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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From brian at freeswitch.org  Sat Feb 13 05:00:15 2016
From: brian at freeswitch.org (Brian West)
Date: Fri, 12 Feb 2016 20:00:15 -0600
Subject: [Freeswitch-users] Get actual duration of the call after hangup
In-Reply-To: <CAD_iGr-M-X6ERmm7=foUPfc_jx6FTzu9Ayy+5YtMSPtW_qWD1w@mail.gmail.com>
References: <CAD_iGr-M-X6ERmm7=foUPfc_jx6FTzu9Ayy+5YtMSPtW_qWD1w@mail.gmail.com>
Message-ID: <CAEJMVkCoHewoJZSWSBR89zDrGfQkVae-ZeFNiEbLNN9J-Rjg+Q@mail.gmail.com>

Parse the xml cdr it will give you more than you want probably.

On Friday, February 12, 2016, Alexandru Covalschi <568691 at gmail.com> wrote:

> Hello folks!
>
> I have a problem - I need to get actual duration of the call - I mean from
> ANSWER to HANGUP, without the time user waited being parked on the bridge.
> The thing is I need to write that info into a custom table which belongs to
> a custom database where I have leg_a uuid.
> How can I achieve that in case of:
> 1. Outoing origination via bridge()
> 2. Connecting two customers which are already inside a freeswitch via
> bridge()
> I've read about mod_cdr_csv but it afaik doesn't allow to connect to a
> custom db.
> Would mod_odbc_cdr allow me to do that? And if yes - can you tell me
> "chan-var-name"?
>
> And don't be afraid - such slow method is not for billing :) Just to
> provide customer more detailed calls history.
>
> Thanks!
> --
> Alexandru Covalschi
> ABRISS-Solutions
> VoIP engineer and system administrator
> phone: +37367398493
> web: http://abs-telecom.com/
>


-- 

*Brian West*
brian at freeswitch.org


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

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 rutu.patel at inextrix.com  Sat Feb 13 07:43:12 2016
From: rutu.patel at inextrix.com (Rutu Patel)
Date: Sat, 13 Feb 2016 10:13:12 +0530
Subject: [Freeswitch-users] Call recording audio too fast
Message-ID: <CAPQet1wQi7nzuuhLPQ=pVXwZ+S8jbPd1NFu0pjUCOioryBE+XA@mail.gmail.com>

Hi All,

I am using record_session application to record the calls but the audio
speed has too fast in recording files.

Below is the code I am using for call recording:

<action application="set" data="RECORD_STEREO=true"/>
<action application="set" data="record_sample_rate=8000"/>
<action application="export" data="execute_on_answer=record_session
$${base_dir}/recordings/${uuid}.wav">

Can anyone please help me to solve the issue with fast audio speed in
recording files ?
Also is it possible to decrease the size of .wav audio file ?
--
Thanks,
Rutu Patel

<http://www.inextrix.com>
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From brian at freeswitch.org  Sat Feb 13 07:55:30 2016
From: brian at freeswitch.org (Brian West)
Date: Fri, 12 Feb 2016 22:55:30 -0600
Subject: [Freeswitch-users] Call recording audio too fast
In-Reply-To: <CAPQet1wQi7nzuuhLPQ=pVXwZ+S8jbPd1NFu0pjUCOioryBE+XA@mail.gmail.com>
References: <CAPQet1wQi7nzuuhLPQ=pVXwZ+S8jbPd1NFu0pjUCOioryBE+XA@mail.gmail.com>
Message-ID: <CAEJMVkA57Z=z4tBFj_6hS6SHXaz4NqTvAycP+Fbr5F73sLgz2w@mail.gmail.com>

Oh great swami nanda, what revision are you running?  We would need a few
more details to know for sure, have you tried the latest master?

On Friday, February 12, 2016, Rutu Patel <rutu.patel at inextrix.com> wrote:

> Hi All,
>
> I am using record_session application to record the calls but the audio
> speed has too fast in recording files.
>
> Below is the code I am using for call recording:
>
> <action application="set" data="RECORD_STEREO=true"/>
> <action application="set" data="record_sample_rate=8000"/>
> <action application="export" data="execute_on_answer=record_session
> $${base_dir}/recordings/${uuid}.wav">
>
> Can anyone please help me to solve the issue with fast audio speed in
> recording files ?
> Also is it possible to decrease the size of .wav audio file ?
> --
> Thanks,
> Rutu Patel
>
> <http://www.inextrix.com>
>


-- 

*Brian West*
brian at freeswitch.org


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

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 rutu.patel at inextrix.com  Sat Feb 13 08:10:00 2016
From: rutu.patel at inextrix.com (Rutu Patel)
Date: Sat, 13 Feb 2016 10:40:00 +0530
Subject: [Freeswitch-users] Call recording audio too fast
In-Reply-To: <CAEJMVkA57Z=z4tBFj_6hS6SHXaz4NqTvAycP+Fbr5F73sLgz2w@mail.gmail.com>
References: <CAPQet1wQi7nzuuhLPQ=pVXwZ+S8jbPd1NFu0pjUCOioryBE+XA@mail.gmail.com>
	<CAEJMVkA57Z=z4tBFj_6hS6SHXaz4NqTvAycP+Fbr5F73sLgz2w@mail.gmail.com>
Message-ID: <CAPQet1ztC500_jJQyodCLvGgFaKYR2jS0rz_TdUkQzkWg1aJ3w@mail.gmail.com>

Hi Brian,

I am using freeswitch-1.6.5 on Debian-8.2.
What further information I should share ?

--
Thanks,
Rutu Patel
<http://www.inextrix.com>

On Sat, Feb 13, 2016 at 10:25 AM, Brian West <brian at freeswitch.org> wrote:

> Oh great swami nanda, what revision are you running?  We would need a few
> more details to know for sure, have you tried the latest master?
>
>
> On Friday, February 12, 2016, Rutu Patel <rutu.patel at inextrix.com> wrote:
>
>> Hi All,
>>
>> I am using record_session application to record the calls but the audio
>> speed has too fast in recording files.
>>
>> Below is the code I am using for call recording:
>>
>> <action application="set" data="RECORD_STEREO=true"/>
>> <action application="set" data="record_sample_rate=8000"/>
>> <action application="export" data="execute_on_answer=record_session
>> $${base_dir}/recordings/${uuid}.wav">
>>
>> Can anyone please help me to solve the issue with fast audio speed in
>> recording files ?
>> Also is it possible to decrease the size of .wav audio file ?
>> --
>> Thanks,
>> Rutu Patel
>>
>> <http://www.inextrix.com>
>>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> 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 brian at freeswitch.org  Sat Feb 13 08:15:52 2016
From: brian at freeswitch.org (Brian West)
Date: Fri, 12 Feb 2016 23:15:52 -0600
Subject: [Freeswitch-users] Call recording audio too fast
In-Reply-To: <CAPQet1ztC500_jJQyodCLvGgFaKYR2jS0rz_TdUkQzkWg1aJ3w@mail.gmail.com>
References: <CAPQet1wQi7nzuuhLPQ=pVXwZ+S8jbPd1NFu0pjUCOioryBE+XA@mail.gmail.com>
	<CAEJMVkA57Z=z4tBFj_6hS6SHXaz4NqTvAycP+Fbr5F73sLgz2w@mail.gmail.com>
	<CAPQet1ztC500_jJQyodCLvGgFaKYR2jS0rz_TdUkQzkWg1aJ3w@mail.gmail.com>
Message-ID: <CAEJMVkDTVqJ0QDQjX+W=wLhFQOFY1vXTan=seJ78etv6RKcuMA@mail.gmail.com>

Can you try 1.6.6 then post logs if a call?

On Friday, February 12, 2016, Rutu Patel <rutu.patel at inextrix.com> wrote:

> Hi Brian,
>
> I am using freeswitch-1.6.5 on Debian-8.2.
> What further information I should share ?
>
> --
> Thanks,
> Rutu Patel
> <http://www.inextrix.com>
>
> On Sat, Feb 13, 2016 at 10:25 AM, Brian West <brian at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','brian at freeswitch.org');>> wrote:
>
>> Oh great swami nanda, what revision are you running?  We would need a few
>> more details to know for sure, have you tried the latest master?
>>
>>
>> On Friday, February 12, 2016, Rutu Patel <rutu.patel at inextrix.com
>> <javascript:_e(%7B%7D,'cvml','rutu.patel at inextrix.com');>> wrote:
>>
>>> Hi All,
>>>
>>> I am using record_session application to record the calls but the audio
>>> speed has too fast in recording files.
>>>
>>> Below is the code I am using for call recording:
>>>
>>> <action application="set" data="RECORD_STEREO=true"/>
>>> <action application="set" data="record_sample_rate=8000"/>
>>> <action application="export" data="execute_on_answer=record_session
>>> $${base_dir}/recordings/${uuid}.wav">
>>>
>>> Can anyone please help me to solve the issue with fast audio speed in
>>> recording files ?
>>> Also is it possible to decrease the size of .wav audio file ?
>>> --
>>> Thanks,
>>> Rutu Patel
>>>
>>> <http://www.inextrix.com>
>>>
>>
>>
>> --
>>
>> *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
>>
>> 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
>> <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


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

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 jprangi at didforsale.com  Sat Feb 13 23:23:38 2016
From: jprangi at didforsale.com (Jai Rangi)
Date: Sat, 13 Feb 2016 12:23:38 -0800
Subject: [Freeswitch-users] Oneway audio issues in freeswitch
In-Reply-To: <CAPfjz3jwe74Vvz1515TQxOJdAYsNuRXA_CTtzTPfG9EjuXXhEA@mail.gmail.com>
References: <CANw+5nWmd_j=eUz7OGoH=5Wb3g43VOS_XF_dZVkuFRMRUtfZ2Q@mail.gmail.com>
	<CABMX_w_NWvQ5ZUdxtYHnua4FD9i4hYDt3YeP-X+43XYi4cb=8A@mail.gmail.com>
	<CAFLo2QzdR1YBLv6-y5hs=juyE5N2uZw+N5dCiyq4S3Hbgkb2eg@mail.gmail.com>
	<CANw+5nUWV3dmJGUtDgLBSaUuGufcXskoYNyj6FJcmd+yRs184g@mail.gmail.com>
	<CAPcNkdfppepyyC+OLTr4OFyhFWhP9MjWR0WUsraU-ABrcJLY1Q@mail.gmail.com>
	<CAPfjz3jwe74Vvz1515TQxOJdAYsNuRXA_CTtzTPfG9EjuXXhEA@mail.gmail.com>
Message-ID: <CAPfjz3g1D+zsiWfmKQzVdviwCH9UX+pM7gLbjT2aoTKFPTmuLQ@mail.gmail.com>

Found the problem, We had this enabled in profiles.
auto-jitterbuffer-msec value=60
Commenting it seems to have fixed the issue. Not sure why this would cause
problem only from one type of phones. Any hint.

Thank you,






*Jai Rangi*
Cebod Technologies LLC dba DIDforSale/Cebod Telecom
O 949-471-0102 <O%20949-471-0101> | C 949-419-7634 <C%20949-742-2666> | F
949-269-0449 / 949-232-1410 | jprangi at didforsale.com  www.cebod.com |
www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA 92626  |






On Sat, Feb 13, 2016 at 9:45 AM, Jai Rangi <jprangi at didforsale.com> wrote:

> Hello Ken,
> Thank for look in this. Attached are debug logs. SIP Traces were not
> molested, except the public IPs were changed. As of writing of this email,
> the issue is isolated to 1.6.x.
> Not sure if anyone else has tested this on latest version. But easy to
> reproduce. Just download grandstream Wave, available to IOS and Andriod and
> try to call any extension directly. Curious to see if any one can come with
> different result.
>
> *Jai Rangi*
> Cebod Technologies LLC dba DIDforSale/Cebod Telecom
> O 949-471-0102 <O%20949-471-0101> | C 949-419-7634 <C%20949-742-2666> | F
> 949-269-0449 / 949-232-1410 | jprangi at didforsale.com  www.cebod.com |
> www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA 92626  |
>
>
>
>
>
>
> On Thu, Feb 11, 2016 at 8:36 AM, Ken Rice <krice at freeswitch.org> wrote:
>
>> without logs of a call doing this at debug level with a complete
>> unmolested sip trace in line its a little hard to speculate whats going on
>> here
>>
>> On Thu, Feb 11, 2016 at 10:28 AM, mohammed shafeeque <shafeeq.v at gmail.com
>> > wrote:
>>
>>> Surprised that no one else experienced this problem. Can anyone give any
>>> hint. Really Dont want to move back to 1.4.x
>>>
>>> On Thu, Feb 11, 2016 at 7:44 AM, Luis Daniel Lucio Quiroz <
>>> luis.daniel.lucio at gmail.com> wrote:
>>>
>>>> As a rule of dumb, try turning on rport
>>>> Le 10 f?vr. 2016 8:55 PM, "?talo Rossi" <italo at freeswitch.org> a
>>>> ?crit :
>>>>
>>>>> You need to look at the sip signaling to see what's going on
>>>>>
>>>>> On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque <
>>>>> shafeeq.v at gmail.com> wrote:
>>>>>
>>>>>> Hello All
>>>>>>
>>>>>> We are getting one way audio issues with some softphones and
>>>>>> grandstream phones behind nat registerd to our freeswitch server.
>>>>>>
>>>>>> Here is scenario:
>>>>>> Grandstream call any extensions (one way audio)
>>>>>> Any extension call Grandstream ( Audio works just fine)
>>>>>>
>>>>>> We have tried multiple softphones and the result is same.
>>>>>>
>>>>>> Everything was working fine with 1.4.18 and 1.6.2. We were having
>>>>>> DTMF issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue
>>>>>> started with an upgrade to freeswitch.
>>>>>>
>>>>>> Any help or hint will be much appreciated.
>>>>>>
>>>>>> Thank you,
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> ?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
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 jprangi at didforsale.com  Sat Feb 13 20:45:43 2016
From: jprangi at didforsale.com (Jai Rangi)
Date: Sat, 13 Feb 2016 09:45:43 -0800
Subject: [Freeswitch-users] Oneway audio issues in freeswitch
In-Reply-To: <CAPcNkdfppepyyC+OLTr4OFyhFWhP9MjWR0WUsraU-ABrcJLY1Q@mail.gmail.com>
References: <CANw+5nWmd_j=eUz7OGoH=5Wb3g43VOS_XF_dZVkuFRMRUtfZ2Q@mail.gmail.com>
	<CABMX_w_NWvQ5ZUdxtYHnua4FD9i4hYDt3YeP-X+43XYi4cb=8A@mail.gmail.com>
	<CAFLo2QzdR1YBLv6-y5hs=juyE5N2uZw+N5dCiyq4S3Hbgkb2eg@mail.gmail.com>
	<CANw+5nUWV3dmJGUtDgLBSaUuGufcXskoYNyj6FJcmd+yRs184g@mail.gmail.com>
	<CAPcNkdfppepyyC+OLTr4OFyhFWhP9MjWR0WUsraU-ABrcJLY1Q@mail.gmail.com>
Message-ID: <CAPfjz3jwe74Vvz1515TQxOJdAYsNuRXA_CTtzTPfG9EjuXXhEA@mail.gmail.com>

Hello Ken,
Thank for look in this. Attached are debug logs. SIP Traces were not
molested, except the public IPs were changed. As of writing of this email,
the issue is isolated to 1.6.x.
Not sure if anyone else has tested this on latest version. But easy to
reproduce. Just download grandstream Wave, available to IOS and Andriod and
try to call any extension directly. Curious to see if any one can come with
different result.

*Jai Rangi*
Cebod Technologies LLC dba DIDforSale/Cebod Telecom
O 949-471-0102 <O%20949-471-0101> | C 949-419-7634 <C%20949-742-2666> | F
949-269-0449 / 949-232-1410 | jprangi at didforsale.com  www.cebod.com |
www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA 92626  |






On Thu, Feb 11, 2016 at 8:36 AM, Ken Rice <krice at freeswitch.org> wrote:

> without logs of a call doing this at debug level with a complete
> unmolested sip trace in line its a little hard to speculate whats going on
> here
>
> On Thu, Feb 11, 2016 at 10:28 AM, mohammed shafeeque <shafeeq.v at gmail.com>
> wrote:
>
>> Surprised that no one else experienced this problem. Can anyone give any
>> hint. Really Dont want to move back to 1.4.x
>>
>> On Thu, Feb 11, 2016 at 7:44 AM, Luis Daniel Lucio Quiroz <
>> luis.daniel.lucio at gmail.com> wrote:
>>
>>> As a rule of dumb, try turning on rport
>>> Le 10 f?vr. 2016 8:55 PM, "?talo Rossi" <italo at freeswitch.org> a ?crit :
>>>
>>>> You need to look at the sip signaling to see what's going on
>>>>
>>>> On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque <
>>>> shafeeq.v at gmail.com> wrote:
>>>>
>>>>> Hello All
>>>>>
>>>>> We are getting one way audio issues with some softphones and
>>>>> grandstream phones behind nat registerd to our freeswitch server.
>>>>>
>>>>> Here is scenario:
>>>>> Grandstream call any extensions (one way audio)
>>>>> Any extension call Grandstream ( Audio works just fine)
>>>>>
>>>>> We have tried multiple softphones and the result is same.
>>>>>
>>>>> Everything was working fine with 1.4.18 and 1.6.2. We were having
>>>>> DTMF issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue
>>>>> started with an upgrade to freeswitch.
>>>>>
>>>>> Any help or hint will be much appreciated.
>>>>>
>>>>> Thank you,
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> ?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
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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2016-02-13 09:22:50.747900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/Master.csv
2016-02-13 09:22:50.747900 [NOTICE] mod_logfile.c:213 New log started.

a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.157920 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/1276 at domain.example.com [a17a5d15-d97e-4c70-b476-bd0ff4ac2e09]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.157920 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at domain.example.com) Running State Change CS_NEW
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.157920 [DEBUG] sofia.c:9248 sofia/internal/1276 at domain.example.com receiving invite from 68.5.94.63:12113 version: 1.6.6  64bit
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.157920 [ALERT] switch_core_media.c:413 Looking for zrtp-hash
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.157920 [ALERT] switch_core_media.c:368 Deciding whether to pass zrtp-hash between legs
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.157920 [ALERT] switch_core_media.c:370 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
2016-02-13 09:23:10.157920 [DEBUG] sofia.c:9415 IP 68.5.94.63 Rejected by acl "domains". Falling back to Digest auth.
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.157920 [DEBUG] switch_core_state_machine.c:492 (sofia/internal/1276 at domain.example.com) State NEW
2016-02-13 09:23:10.157920 [DEBUG] sofia.c:2147 detaching session a17a5d15-d97e-4c70-b476-bd0ff4ac2e09
2016-02-13 09:23:10.387894 [DEBUG] sofia.c:2255 Re-attaching to session a17a5d15-d97e-4c70-b476-bd0ff4ac2e09
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.397894 [DEBUG] sofia.c:9248 sofia/internal/1276 at domain.example.com receiving invite from 68.5.94.63:12113 version: 1.6.6  64bit
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.397894 [ALERT] switch_core_media.c:413 Looking for zrtp-hash
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.397894 [ALERT] switch_core_media.c:368 Deciding whether to pass zrtp-hash between legs
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.397894 [ALERT] switch_core_media.c:370 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
2016-02-13 09:23:10.397894 [DEBUG] sofia.c:9415 IP 68.5.94.63 Rejected by acl "domains". Falling back to Digest auth.
2016-02-13 09:23:10.397894 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f41ac03e390 Connected.
2016-02-13 09:23:10.407800 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f41ac03e390 released.
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] sofia.c:10549 Setting NAT mode based on nat.auto
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] sofia.c:6760 Channel sofia/internal/1276 at domain.example.com entering state [received][100]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] sofia.c:6770 Remote SDP:
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 v=0
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 o=1276 21012 1 IN IP4 192.168.1.168
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 s=-
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 c=IN IP4 192.168.1.168
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 t=0 0
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 m=audio 8002 RTP/AVP 0 18 8 96
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=rtpmap:0 PCMU/8000
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=rtpmap:18 G729/8000
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=rtpmap:8 PCMA/8000
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=rtpmap:96 telephone-event/8000
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=ptime:20
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] sofia.c:7125 (sofia/internal/1276 at domain.example.com) State Change CS_NEW -> CS_INIT
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at domain.example.com) Running State Change CS_INIT
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/1276 at domain.example.com) State INIT
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] mod_sofia.c:88 sofia/internal/1276 at domain.example.com SOFIA INIT
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_core_state_machine.c:40 sofia/internal/1276 at domain.example.com Standard INIT
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/1276 at domain.example.com) State Change CS_INIT -> CS_ROUTING
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/1276 at domain.example.com) State INIT going to sleep
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at domain.example.com) Running State Change CS_ROUTING
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_channel.c:2247 (sofia/internal/1276 at domain.example.com) Callstate Change DOWN -> RINGING
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/1276 at domain.example.com) State ROUTING
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] mod_sofia.c:141 sofia/internal/1276 at domain.example.com SOFIA ROUTING
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_core_state_machine.c:166 sofia/internal/1276 at domain.example.com Standard ROUTING
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [INFO] mod_dialplan_xml.c:637 Processing Kunal Mittal <1276>->142 in context public
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [public->limit_exceeded] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [limit_exceeded] destination_number(142) =~ /^limit_exceeded$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [public->from_testopensip] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [from_testopensip] network_addr(68.5.94.63) =~ /^209\.216\.15\.18$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [public->from_productionopensip] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [from_productionopensip] network_addr(68.5.94.63) =~ /^209\.216\.2\.222$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [public->from_registeredusers] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Absolute Condition [from_registeredusers]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action set(dialed_domain=${sip_to_host})
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action set(domain_name=${sip_to_host})
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action set(company_name=${sip_to_host})
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action export(caller-id-in-from=true)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action export(sip_cid_type=none)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action log(Public IP is ${bind_server_ip} Domain is ${domain_name} Dialed domain is ${dialed_domain})
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action lua(didhandle.lua ${destination_number})
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action log(INFO public Local URI var_name chan var is ${domain_name} Dialed domain is ${dialed_domain})
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_core_state_machine.c:216 (sofia/internal/1276 at domain.example.com) State Change CS_ROUTING -> CS_EXECUTE
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/1276 at domain.example.com) State ROUTING going to sleep
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at domain.example.com) Running State Change CS_EXECUTE
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_core_state_machine.c:539 (sofia/internal/1276 at domain.example.com) State EXECUTE
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] mod_sofia.c:196 sofia/internal/1276 at domain.example.com SOFIA EXECUTE
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_core_state_machine.c:258 sofia/internal/1276 at domain.example.com Standard EXECUTE
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com set(dialed_domain=domain.example.com)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1276 at domain.example.com [dialed_domain]=[domain.example.com]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com set(domain_name=domain.example.com)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1276 at domain.example.com [domain_name]=[domain.example.com]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com set(company_name=domain.example.com)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1276 at domain.example.com [company_name]=[domain.example.com]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com export(caller-id-in-from=true)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [caller-id-in-from]=[true]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com export(sip_cid_type=none)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [sip_cid_type]=[none]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com log(Public IP is 219.206.20.22 Domain is domain.example.com Dialed domain is domain.example.com)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] mod_dptools.c:1692 IP is 219.206.20.22 Domain is domain.example.com Dialed domain is domain.example.com
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com lua(didhandle.lua 142)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
2016-02-13 09:23:10.407800 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f41ac03e390 Connected.
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com export(domain_name=domain.example.com)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [domain_name]=[domain.example.com]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com export(skip_cdr_causes=LOSE_RACE)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [skip_cdr_causes]=[LOSE_RACE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [INFO] switch_cpp.cpp:1284 RPID is  dialing number is 1276
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com export(accountcode=17)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [accountcode]=[17]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [INFO] switch_cpp.cpp:1284 String length is less that 7 should be extension
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com set(process_cdr=a_only)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1276 at domain.example.com [process_cdr]=[a_only]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_ivr.c:2085 (sofia/internal/1276 at domain.example.com) State Change CS_EXECUTE -> CS_ROUTING
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_ivr.c:2090 sofia/internal/1276 at domain.example.com receive message [TRANSFER]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [NOTICE] switch_ivr.c:2092 Transfer sofia/internal/1276 at domain.example.com to XML[142 at default]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_cpp.cpp:898 transfer result: 0
2016-02-13 09:23:10.417792 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f41ac03e390 released.
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_cpp.cpp:1103 sofia/internal/1276 at domain.example.com destroy/unlink session from object
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_core_state_machine.c:539 (sofia/internal/1276 at domain.example.com) State EXECUTE going to sleep
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at domain.example.com) Running State Change CS_ROUTING
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/1276 at domain.example.com) State ROUTING
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] mod_sofia.c:141 sofia/internal/1276 at domain.example.com SOFIA ROUTING
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_core_state_machine.c:166 sofia/internal/1276 at domain.example.com Standard ROUTING
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [INFO] mod_dialplan_xml.c:637 Processing Kunal Mittal <1276>->142 in context default
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->longdistance] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [longdistance] destination_number(142) =~ /^\d{10,15}$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->unloop] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->freeswitch_public_conf_via_sip] continue=false
2016-02-13 09:23:10.417792 [ERR] switch_regex.c:104 COMPILE ERROR: 1 [nothing to repeat][^*9(888|8888|1616|3232)$]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [freeswitch_public_conf_via_sip] destination_number(142) =~ /^*9(888|8888|1616|3232)$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->tod_example] continue=true
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Date/TimeMatch (FAIL) [tod_example] break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->holiday_example] continue=true
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Date/TimeMatch (FAIL) [holiday_example] break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->global-intercept] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [global-intercept] destination_number(142) =~ /^\*886$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->group-intercept] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [group-intercept] destination_number(142) =~ /^\*8$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->intercept-ext] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [intercept-ext] destination_number(142) =~ /^\*\*(\d+)$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->redial] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [redial] destination_number(142) =~ /^\*(redial|870)$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->global] continue=true
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (PASS) [global] ${endpoint_disposition}(DELAYED NEGOTIATION) =~ /^(DELAYED NEGOTIATION)/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [global] ${switch_r_sdp}(v=0
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 o=1276 21012 1 IN IP4 192.168.1.168
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 s=-
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 c=IN IP4 192.168.1.168
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 t=0 0
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 m=audio 8002 RTP/AVP 0 18 8 96
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=rtpmap:0 PCMU/8000
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=rtpmap:18 G729/8000
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=rtpmap:8 PCMA/8000
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=rtpmap:96 telephone-event/8000
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=ptime:20
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 ) =~ /(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)/ break=never
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Absolute Condition [global]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action hash(insert/${domain_name}-last_dial/global/${uuid})
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->snom-demo-2] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [snom-demo-2] destination_number(142) =~ /^9001$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->snom-demo-1] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [snom-demo-1] destination_number(142) =~ /^9000$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->eavesdrop] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [eavesdrop] destination_number(142) =~ /^\*88(\d{3,5})$|^\*0(.*)$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->eavesdrop] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [eavesdrop] destination_number(142) =~ /^\*779$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->call_return] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [call_return] destination_number(142) =~ /^\*69$|^\*869$|^lcr$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->del-number] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [del-number] destination_number(142) =~ /^\*60$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->add-number] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [add-number] destination_number(142) =~ /^\*61$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->check-number] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [check-number] destination_number(142) =~ /^\*62$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->enable-time-condition] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [enable-time-condition] destination_number(142) =~ /^\*63$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->disable-time-condition] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [disable-time-condition] destination_number(142) =~ /^\*64$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->del-group] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [del-group] destination_number(142) =~ /^\*80$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->add-group] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [add-group] destination_number(142) =~ /^\*81$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->call-group-simo] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [call-group-simo] destination_number(142) =~ /^\*82(\d{2})$|^82(\d{2})$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->call-group-order] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [call-group-order] destination_number(142) =~ /^\*83(\d{2})$|^83(\d{2})$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->extension-intercom] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [extension-intercom] destination_number(142) =~ /^\*84(\d{3,5})$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->user_recodring_enabled] continue=true
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (PASS) [user_recodring_enabled] ${recording}(off) =~ /^(on|off)$/ break=on-true
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->user_callscreening] continue=true
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [user_callscreening] caller_id_number(1276) =~ /^\d{10,15}$/ break=on-true
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (PASS) [user_callscreening] ${callscreening}(off) =~ /^(on|off)$/ break=on-true
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->Check IVR-based CF] continue=true
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (PASS) [Check IVR-based CF] destination_number(142) =~ /^(\d+)$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action set(dialed_number=142) INLINE
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com set(dialed_number=142)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1276 at domain.example.com [dialed_number]=[142]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action set(cf_target=${db(select/${domain_name}-CF/142)}) INLINE
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com set(cf_target=)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1276 at domain.example.com [cf_target]=[UNDEF]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [Check IVR-based CF] ${recording}(off) =~ /^on$/ break=never
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [Check IVR-based CF] ${cf_target}() =~ /^\d{3,5}$/ break=never
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [Check IVR-based CF] ${cf_target}() =~ /^\d{10,15}$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->CF from my station] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [CF from my station] destination_number(142) =~ /^\*72$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->CF cancel from my station] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [CF cancel from my station] destination_number(142) =~ /^\*73$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->CF from any station] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [CF from any station] destination_number(142) =~ /^\*76$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->CF cancel from any station] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [CF cancel from any station] destination_number(142) =~ /^\*77$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->NACF from my station] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [NACF from my station] destination_number(142) =~ /^\*74$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->NACF cancel from my station] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [NACF cancel from my station] destination_number(142) =~ /^\*75$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->Add_Member_To_Queue] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [Add_Member_To_Queue] destination_number(142) =~ /^\*51$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->Delete_Member_From_Queue] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [Delete_Member_From_Queue] destination_number(142) =~ /^\*52$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->Pickup_Call_From_Queue] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [Pickup_Call_From_Queue] destination_number(142) =~ /^\*53$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->dial-by-name] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [dial-by-name] destination_number(142) =~ /^\*40$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->conference_withpin] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [conference_withpin] destination_number(142) =~ /^\*3560$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->Voicemail_check] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [Voicemail_check] caller_id_number(1276) =~ /^(142)$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->check_availability] continue=true
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [check_availability] destination_number(142) =~ /^hkjhk(1[0-9][0-9][0-9]|[1-2][0-9][0-9])$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->is_check_availability] continue=true
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [is_check_availability] available_to_call() =~ /true/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->osbridge] continue=true
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (PASS) [osbridge] destination_number(142) =~ /(^\d{3,5})$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action set(proxy_media=false)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action set(continue_on_fail=true)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action export(dialed_extension=142) INLINE
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com export(dialed_extension=142)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [dialed_extension]=[142]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action export(domain_name=${domain_name})
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action set(accountcode=${user_data(${dialed_extension}@${domain_name} var accountcode)})
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action export(available_to_call=true) INLINE
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com export(available_to_call=true)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [available_to_call]=[true]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action log(Extensions ${dialed_extension} is ${available_to_call})
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action set(nacf_target=${db(select/${domain_name}-NACF/142)}) INLINE
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com set(nacf_target=)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1276 at domain.example.com [nacf_target]=[UNDEF]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [osbridge] ${recording}(off) =~ /^on$/ break=never
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (PASS) [osbridge] ${available_to_call}(true) =~ /^true$/ break=never
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action bind_meta_app(1 b s execute_extension::dx XML features)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/var/lib/freeswitch/recordings/${domain_name}/${strftime(%Y-%m-%d-%H-%M-%S)}.${uuid}.wav)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action bind_meta_app(3 b s execute_extension::cf XML features)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action bind_meta_app(4 b s execute_extension::attented_xfer XML default)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action bind_meta_app(5 b s execute_extension::attented_xfer_to_vm XML default)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action bind_meta_app(6 b s execute_extension::blind_xfer_to XML default)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action bind_meta_app(9 b s execute_extension::call_park XML default)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action export(sip_contact_user=${sip_from_user})
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action set(ringback=${us-ring})
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action set(transfer_ringback=local_stream://moh)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action set(call_timeout=35)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action set(hangup_after_bridge=true)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action bridge(user/${dialed_extension}@${domain_name})
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action sleep(1000)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [osbridge] ${nacf_target}() =~ /^\d{3,7}$/ break=never
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [osbridge] ${nacf_target}() =~ /^\d{10,15}$/ break=never
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Absolute Condition [osbridge]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action answer()
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action sleep(500)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action voicemail(default ${domain_name} ${dialed_extension})
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->Local_Extension_VM] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [Local_Extension_VM] destination_number(142) =~ /^vm-(\d{3,5})$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->attented_xfer_to_vm] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [attented_xfer_to_vm] destination_number(142) =~ /^attented_xfer_to_vm$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->attented_xfer] continue=true
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [attented_xfer] destination_number(142) =~ /^attented_xfer$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->blind_xfer_to] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [blind_xfer_to] destination_number(142) =~ /^blind_xfer_to$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->Local_Extension] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [Local_Extension] destination_number(142) =~ /^disable-(1[0-9][0-9][0-9]|[1-2][0-9][0-9])$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->Local_Extension_Skinny] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [Local_Extension_Skinny] destination_number(142) =~ /^(11[01][0-9])$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->park-in] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [park-in] destination_number(142) =~ /^call_park$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->park-out] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [park-out] destination_number(142) =~ /^\*(85\d\d)$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->vmain] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [vmain] destination_number(142) =~ /^vmain$|^\*98$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->sip_uri] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [sip_uri] destination_number(142) =~ /^sip:(.*)$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->local_uri] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [local_uri] destination_number(142) =~ /^(local_.*)$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->nb_conferences] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [nb_conferences] destination_number(142) =~ /^\*(30\d{2,4})$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->wb_conferences] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [wb_conferences] destination_number(142) =~ /^\*(31\d{2})$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->uwb_conferences] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [uwb_conferences] destination_number(142) =~ /^\*(32\d{2})$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->cdquality_conferences] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [cdquality_conferences] destination_number(142) =~ /^\*(33\d{2})$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->mad_boss_intercom] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [mad_boss_intercom] destination_number(142) =~ /^0911$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->mad_boss_intercom] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [mad_boss_intercom] destination_number(142) =~ /^0912$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->mad_boss] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [mad_boss] destination_number(142) =~ /^0913$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->queupark] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [queupark] destination_number(142) =~ /^\*(5902)$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->queueunpark] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [queueunpark] destination_number(142) =~ /^\*(5903)$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->park] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [park] destination_number(142) =~ /^\*5900$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->unpark] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [unpark] destination_number(142) =~ /^\*(5901)$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->valet_park] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [valet_park] destination_number(142) =~ /^(6000)$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->valet_park] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [valet_park] destination_number(142) =~ /^(60\d[1-9])$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->park] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [park] destination_number(142) =~ /park\+(\d+)/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->unpark] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [unpark] destination_number(142) =~ /^parking$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->park] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [park] destination_number(142) =~ /callpark/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->unpark] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [unpark] destination_number(142) =~ /pickup/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->wait] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [wait] destination_number(142) =~ /^wait$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->fax_receive] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [fax_receive] destination_number(142) =~ /^fax.*$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->fax_transmit] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [fax_transmit] destination_number(142) =~ /^9179$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->ringback_180] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [ringback_180] destination_number(142) =~ /^9180$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->ringback_183_uk_ring] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [ringback_183_uk_ring] destination_number(142) =~ /^\*9181$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->ringback_183_music_ring] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [ringback_183_music_ring] destination_number(142) =~ /^\*9182$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->ringback_post_answer_uk_ring] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [ringback_post_answer_uk_ring] destination_number(142) =~ /^\*9183$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->ringback_post_answer_music] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [ringback_post_answer_music] destination_number(142) =~ /^\*9184$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->show_info] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [show_info] destination_number(142) =~ /^\*9192$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->video_record] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [video_record] destination_number(142) =~ /^\*9193$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->video_playback] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [video_playback] destination_number(142) =~ /^\*9194$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->delay_echo] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [delay_echo] destination_number(142) =~ /^\*9195$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->echo] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [echo] destination_number(142) =~ /^\*9196$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->milliwatt] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [milliwatt] destination_number(142) =~ /^\*9197$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->tone_stream] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [tone_stream] destination_number(142) =~ /^\*9198$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->zrtp_enrollement] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [zrtp_enrollement] destination_number(142) =~ /^\*9787$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->hold_music] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [hold_music] destination_number(142) =~ /^\*9664$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->laugh break] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [laugh break] destination_number(142) =~ /^\*9386$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->101] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [101] destination_number(142) =~ /^101$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->pizza_demo] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [pizza_demo] destination_number(142) =~ /^(pizza|74992)$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->Talking Clock Time] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [Talking Clock Time] destination_number(142) =~ /^9170$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->Talking Clock Date] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [Talking Clock Date] destination_number(142) =~ /^9171$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->Talking Clock Date and Time] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [Talking Clock Date and Time] destination_number(142) =~ /^9172$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->local.example.com] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [local.example.com] ${toll_allow}() =~ /local/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->domestic.example.com] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [domestic.example.com] ${toll_allow}() =~ /domestic/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->international.example.com] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [international.example.com] ${toll_allow}() =~ /international/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->conference] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [conference] destination_number(142) =~ /^112233$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->didforsale_ivr] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [didforsale_ivr] destination_number(142) =~ /5566/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->domian_ivr] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [domian_ivr] destination_number(142) =~ /ivr(1949.*)$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->enum] continue=false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (PASS) [enum] ${module_exists(mod_enum)}(true) =~ /true/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (PASS) [enum] destination_number(142) =~ /^(.*)$/ break=on-false
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action transfer(142 enum)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_core_state_machine.c:216 (sofia/internal/1276 at domain.example.com) State Change CS_ROUTING -> CS_EXECUTE
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/1276 at domain.example.com) State ROUTING going to sleep
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at domain.example.com) Running State Change CS_EXECUTE
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_core_state_machine.c:539 (sofia/internal/1276 at domain.example.com) State EXECUTE
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] mod_sofia.c:196 sofia/internal/1276 at domain.example.com SOFIA EXECUTE
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_core_state_machine.c:258 sofia/internal/1276 at domain.example.com Standard EXECUTE
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com hash(insert/domain.example.com-spymap/1276/a17a5d15-d97e-4c70-b476-bd0ff4ac2e09)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com hash(insert/domain.example.com-last_dial/1276/142)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com hash(insert/domain.example.com-last_dial/global/a17a5d15-d97e-4c70-b476-bd0ff4ac2e09)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com export(RFC2822_DATE=Sat, 13 Feb 2016 09:23:10 -0800)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [RFC2822_DATE]=[Sat, 13 Feb 2016 09:23:10 -0800]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com set(proxy_media=false)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1276 at domain.example.com [proxy_media]=[false]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com set(continue_on_fail=true)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1276 at domain.example.com [continue_on_fail]=[true]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com export(domain_name=domain.example.com)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [domain_name]=[domain.example.com]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
2016-02-13 09:23:10.427795 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f41ac03e390 Connected.
2016-02-13 09:23:10.427795 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f41ac03e390 released.
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com set(accountcode=17)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1276 at domain.example.com [accountcode]=[17]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com log(Extensions 142 is true)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] mod_dptools.c:1692 142 is true
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com bind_meta_app(1 b s execute_extension::dx XML features)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *1 execute_extension::dx XML features
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com bind_meta_app(2 b s record_session::/usr/local/freeswitch/var/lib/freeswitch/recordings/domain.example.com/2016-02-13-09-23-10.a17a5d15-d97e-4c70-b476-bd0ff4ac2e09.wav)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *2 record_session::/usr/local/freeswitch/var/lib/freeswitch/recordings/domain.example.com/2016-02-13-09-23-10.a17a5d15-d97e-4c70-b476-bd0ff4ac2e09.wav
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com bind_meta_app(3 b s execute_extension::cf XML features)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *3 execute_extension::cf XML features
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com bind_meta_app(4 b s execute_extension::attented_xfer XML default)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *4 execute_extension::attented_xfer XML default
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com bind_meta_app(5 b s execute_extension::attented_xfer_to_vm XML default)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *5 execute_extension::attented_xfer_to_vm XML default
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com bind_meta_app(6 b s execute_extension::blind_xfer_to XML default)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *6 execute_extension::blind_xfer_to XML default
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com bind_meta_app(9 b s execute_extension::call_park XML default)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *9 execute_extension::call_park XML default
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com export(sip_contact_user=1276)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [sip_contact_user]=[1276]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com set(ringback=%(2000,4000,440,480))
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1276 at domain.example.com [ringback]=[%(2000,4000,440,480)]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com set(transfer_ringback=local_stream://moh)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1276 at domain.example.com [transfer_ringback]=[local_stream://moh]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com set(call_timeout=35)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1276 at domain.example.com [call_timeout]=[35]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com set(hangup_after_bridge=true)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1276 at domain.example.com [hangup_after_bridge]=[true]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com bridge(user/142 at domain.example.com)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [domain_name]=[domain.example.com] to event
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [caller-id-in-from]=[true] to event
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [sip_cid_type]=[none] to event
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [domain_name]=[domain.example.com] to event
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [skip_cdr_causes]=[LOSE_RACE] to event
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [accountcode]=[17] to event
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [dialed_extension]=[142] to event
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [available_to_call]=[true] to event
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [RFC2822_DATE]=[Sat, 13 Feb 2016 09:23:10 -0800] to event
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [domain_name]=[domain.example.com] to event
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [sip_contact_user]=[1276] to event
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_ivr_originate.c:2128 Parsing global variables
2016-02-13 09:23:10.427795 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f41ac03e390 Connected.
2016-02-13 09:23:10.427795 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f41ac03e390 released.
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [domain_name]=[domain.example.com] to event
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [caller-id-in-from]=[true] to event
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [sip_cid_type]=[none] to event
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [domain_name]=[domain.example.com] to event
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [skip_cdr_causes]=[LOSE_RACE] to event
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [accountcode]=[17] to event
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [dialed_extension]=[142] to event
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [available_to_call]=[true] to event
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [RFC2822_DATE]=[Sat, 13 Feb 2016 09:23:10 -0800] to event
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [domain_name]=[domain.example.com] to event
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [sip_contact_user]=[1276] to event
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_ivr_originate.c:2128 Parsing global variables
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.427795 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/142 at 192.168.33.116:49383 [d9770cc8-1742-40cd-876f-3b5f50733449]
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.427795 [DEBUG] mod_sofia.c:4776 (sofia/internal/142 at 192.168.33.116:49383) State Change CS_NEW -> CS_INIT
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at 192.168.33.116:49383) Running State Change CS_INIT
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/142 at 192.168.33.116:49383) State INIT
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] mod_sofia.c:88 sofia/internal/142 at 192.168.33.116:49383 SOFIA INIT
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] sofia_glue.c:1228 sip:142 at 73.170.102.43:49383 Setting proxy route to sofia/internal/142 at 192.168.33.116:49383
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] sofia_glue.c:1257 sofia/internal/142 at 192.168.33.116:49383 sending invite version: 1.6.6  64bit
d9770cc8-1742-40cd-876f-3b5f50733449 Local SDP:
d9770cc8-1742-40cd-876f-3b5f50733449 v=0
d9770cc8-1742-40cd-876f-3b5f50733449 o=FreeSWITCH 1455355946 1455355947 IN IP4 219.206.20.22
d9770cc8-1742-40cd-876f-3b5f50733449 s=FreeSWITCH
d9770cc8-1742-40cd-876f-3b5f50733449 c=IN IP4 219.206.20.22
d9770cc8-1742-40cd-876f-3b5f50733449 t=0 0
d9770cc8-1742-40cd-876f-3b5f50733449 m=audio 28244 RTP/AVP 0 18 8 101 13
d9770cc8-1742-40cd-876f-3b5f50733449 a=rtpmap:0 PCMU/8000
d9770cc8-1742-40cd-876f-3b5f50733449 a=rtpmap:18 G729/8000
d9770cc8-1742-40cd-876f-3b5f50733449 a=rtpmap:8 PCMA/8000
d9770cc8-1742-40cd-876f-3b5f50733449 a=rtpmap:101 telephone-event/8000
d9770cc8-1742-40cd-876f-3b5f50733449 a=fmtp:101 0-16
d9770cc8-1742-40cd-876f-3b5f50733449 a=rtpmap:13 CN/8000
d9770cc8-1742-40cd-876f-3b5f50733449 a=ptime:20
d9770cc8-1742-40cd-876f-3b5f50733449 a=sendrecv
d9770cc8-1742-40cd-876f-3b5f50733449 
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] switch_core_state_machine.c:40 sofia/internal/142 at 192.168.33.116:49383 Standard INIT
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/142 at 192.168.33.116:49383) State Change CS_INIT -> CS_ROUTING
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/142 at 192.168.33.116:49383) State INIT going to sleep
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at 192.168.33.116:49383) Running State Change CS_ROUTING
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] sofia.c:6760 Channel sofia/internal/142 at 192.168.33.116:49383 entering state [calling][0]
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/142 at 192.168.33.116:49383) State ROUTING
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] mod_sofia.c:141 sofia/internal/142 at 192.168.33.116:49383 SOFIA ROUTING
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/142 at 192.168.33.116:49383) State Change CS_ROUTING -> CS_CONSUME_MEDIA
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/142 at 192.168.33.116:49383) State ROUTING going to sleep
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at 192.168.33.116:49383) Running State Change CS_CONSUME_MEDIA
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] switch_core_state_machine.c:551 (sofia/internal/142 at 192.168.33.116:49383) State CONSUME_MEDIA
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] switch_core_state_machine.c:551 (sofia/internal/142 at 192.168.33.116:49383) State CONSUME_MEDIA going to sleep
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [ALERT] switch_core_state_machine.c:590 sofia/internal/142 at 192.168.33.116:49383 session thread sleep state: CS_CONSUME_MEDIA!
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.507868 [ALERT] switch_core_state_machine.c:594 sofia/internal/142 at 192.168.33.116:49383 session thread wake state: CS_CONSUME_MEDIA!
2016-02-13 09:23:10.507868 [ALERT] sofia.c:1214 sofia/internal/142 at 192.168.33.116:49383 Same Callee ID "Outbound Call" <142>
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.507868 [DEBUG] sofia.c:6760 Channel sofia/internal/142 at 192.168.33.116:49383 entering state [proceeding][180]
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.507868 [NOTICE] sofia.c:6862 Ring-Ready sofia/internal/142 at 192.168.33.116:49383!
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.507868 [DEBUG] switch_channel.c:3340 (sofia/internal/142 at 192.168.33.116:49383) Callstate Change DOWN -> RINGING
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.507868 [ALERT] sofia.c:6862 sofia/internal/142 at 192.168.33.116:49383 receive message [RING_EVENT]
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.507868 [ALERT] switch_core_state_machine.c:590 sofia/internal/142 at 192.168.33.116:49383 session thread sleep state: CS_CONSUME_MEDIA!
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [ALERT] switch_ivr_originate.c:1216 sofia/internal/1276 at domain.example.com receive message [PROGRESS]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [INFO] switch_ivr_originate.c:1216 Sending early media
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [ALERT] switch_core_media.c:413 Looking for zrtp-hash
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [ALERT] switch_core_media.c:368 Deciding whether to pass zrtp-hash between legs
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [ALERT] switch_core_media.c:376 Found peer channel; propagating zrtp-hash if set
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [ALERT] switch_core_media.c:303 Deciding whether to pass zrtp-hash between a-leg and b-leg
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [ALERT] switch_core_media.c:303 Deciding whether to pass zrtp-hash between a-leg and b-leg
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G729:18:8000:20:8000:1]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [G729:18:8000:20:8000:1]/[PCMU:0:8000:20:64000:1]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [G729:18:8000:20:8000:1]/[PCMA:8:8000:20:64000:1]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [G729:18:8000:20:8000:1]/[G729:18:8000:20:8000:1]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [G729:18:8000:20:8000:1] ++++ is saved as a match
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [G729:18:8000:20:8000:1]/[GSM:3:8000:20:13200:1]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G729:18:8000:20:8000:1]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4077 Set telephone-event payload to 96 at 8000
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:2906 Set Codec sofia/internal/1276 at domain.example.com PCMU/8000 20 ms 160 samples 64000 bits 1 channels
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_codec.c:111 sofia/internal/1276 at domain.example.com Original read codec set to PCMU:0
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4429 Set telephone-event payload to 96 at 8000
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4485 sofia/internal/1276 at domain.example.com Set 2833 dtmf send payload to 96 recv payload to 96
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:6033 AUDIO RTP [sofia/internal/1276 at domain.example.com] 219.206.20.22 port 26876 -> 192.168.1.168 port 8002 codec: 0 ms: 20
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_rtp.c:3802 Starting timer [soft] 160 bytes per 20ms
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:1939 Setting Jitterbuffer to 60ms (3 frames) (50 max frames)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:6332 sofia/internal/1276 at domain.example.com Set 2833 dtmf send payload to 96
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:6339 sofia/internal/1276 at domain.example.com Set 2833 dtmf receive payload to 96
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:6362 sofia/internal/1276 at domain.example.com Set rtp dtmf delay to 60
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1276 at domain.example.com!
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_channel.c:3468 (sofia/internal/1276 at domain.example.com) Callstate Change RINGING -> EARLY
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [ALERT] sofia_media.c:92 sofia/internal/1276 at domain.example.com receive message [PROGRESS_EVENT]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] mod_sofia.c:2330 Ring SDP:
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 v=0
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 o=FreeSWITCH 1455357314 1455357315 IN IP4 219.206.20.22
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 s=FreeSWITCH
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 c=IN IP4 219.206.20.22
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 t=0 0
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 m=audio 26876 RTP/AVP 0 96
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=rtpmap:0 PCMU/8000
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=rtpmap:96 telephone-event/8000
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=fmtp:96 0-16
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=ptime:20
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=sendrecv
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_ivr_originate.c:1274 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_codec.c:221 sofia/internal/1276 at domain.example.com Push codec L16:100
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_ivr_originate.c:1343 Play Ringback Tone [%(2000,4000,440,480)]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.537870 [DEBUG] sofia.c:6760 Channel sofia/internal/1276 at domain.example.com entering state [early][183]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.537870 [ALERT] switch_channel.c:3507 sofia/internal/1276 at domain.example.com receive message [AUDIO_SYNC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.537870 [ALERT] switch_core_io.c:1135 sofia/internal/1276 at domain.example.com receive message [TRANSCODING_NECESSARY]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.827958 [INFO] switch_rtp.c:6616 Auto Changing audio port from 192.168.1.168:8002 to 68.5.94.63:8002
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:13.517891 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at domain.example.com Got: audio seq 57074 but expected: 57072 lost: 2
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.537897 [ALERT] switch_core_state_machine.c:594 sofia/internal/142 at 192.168.33.116:49383 session thread wake state: CS_CONSUME_MEDIA!
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.537897 [ALERT] switch_core_media.c:413 Looking for zrtp-hash
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.537897 [ALERT] switch_core_media.c:368 Deciding whether to pass zrtp-hash between legs
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.537897 [ALERT] switch_core_media.c:376 Found peer channel; propagating zrtp-hash if set
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.537897 [ALERT] switch_core_media.c:303 Deciding whether to pass zrtp-hash between a-leg and b-leg
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.537897 [ALERT] switch_core_media.c:303 Deciding whether to pass zrtp-hash between a-leg and b-leg
2016-02-13 09:23:14.547896 [ALERT] sofia.c:1214 sofia/internal/142 at 192.168.33.116:49383 Same Callee ID "Outbound Call" <142>
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] sofia.c:6760 Channel sofia/internal/142 at 192.168.33.116:49383 entering state [completing][200]
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] sofia.c:6770 Remote SDP:
d9770cc8-1742-40cd-876f-3b5f50733449 v=0
d9770cc8-1742-40cd-876f-3b5f50733449 o=142 8000 8000 IN IP4 192.168.33.116
d9770cc8-1742-40cd-876f-3b5f50733449 s=SIP Call
d9770cc8-1742-40cd-876f-3b5f50733449 c=IN IP4 192.168.33.116
d9770cc8-1742-40cd-876f-3b5f50733449 t=0 0
d9770cc8-1742-40cd-876f-3b5f50733449 m=audio 12552 RTP/AVP 0 8 101
d9770cc8-1742-40cd-876f-3b5f50733449 a=rtpmap:0 PCMU/8000
d9770cc8-1742-40cd-876f-3b5f50733449 a=rtpmap:8 PCMA/8000
d9770cc8-1742-40cd-876f-3b5f50733449 a=rtpmap:101 telephone-event/8000
d9770cc8-1742-40cd-876f-3b5f50733449 a=fmtp:101 0-15
d9770cc8-1742-40cd-876f-3b5f50733449 a=ptime:20
d9770cc8-1742-40cd-876f-3b5f50733449 
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [ALERT] switch_core_state_machine.c:590 sofia/internal/142 at 192.168.33.116:49383 session thread sleep state: CS_CONSUME_MEDIA!
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [ALERT] switch_core_state_machine.c:594 sofia/internal/142 at 192.168.33.116:49383 session thread wake state: CS_CONSUME_MEDIA!
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] sofia.c:6760 Channel sofia/internal/142 at 192.168.33.116:49383 entering state [ready][200]
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [ALERT] switch_core_media.c:413 Looking for zrtp-hash
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [ALERT] switch_core_media.c:368 Deciding whether to pass zrtp-hash between legs
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [ALERT] switch_core_media.c:376 Found peer channel; propagating zrtp-hash if set
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [ALERT] switch_core_media.c:303 Deciding whether to pass zrtp-hash between a-leg and b-leg
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [ALERT] switch_core_media.c:303 Deciding whether to pass zrtp-hash between a-leg and b-leg
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G729:18:8000:20:8000:1]
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G729:18:8000:20:8000:1]
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:4077 Set telephone-event payload to 101 at 8000
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:2906 Set Codec sofia/internal/142 at 192.168.33.116:49383 PCMU/8000 20 ms 160 samples 64000 bits 1 channels
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_codec.c:111 sofia/internal/142 at 192.168.33.116:49383 Original read codec set to PCMU:0
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:4429 Set telephone-event payload to 101 at 8000
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:4485 sofia/internal/142 at 192.168.33.116:49383 Set 2833 dtmf send payload to 101 recv payload to 101
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:6033 AUDIO RTP [sofia/internal/142 at 192.168.33.116:49383] 219.206.20.22 port 28244 -> 192.168.33.116 port 12552 codec: 0 ms: 20
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_rtp.c:3802 Starting timer [soft] 160 bytes per 20ms
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:1939 Setting Jitterbuffer to 60ms (3 frames) (50 max frames)
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:6332 sofia/internal/142 at 192.168.33.116:49383 Set 2833 dtmf send payload to 101
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:6339 sofia/internal/142 at 192.168.33.116:49383 Set 2833 dtmf receive payload to 101
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:6362 sofia/internal/142 at 192.168.33.116:49383 Set rtp dtmf delay to 60
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [NOTICE] sofia.c:7724 Channel [sofia/internal/142 at 192.168.33.116:49383] has been answered
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_channel.c:3767 (sofia/internal/142 at 192.168.33.116:49383) Callstate Change RINGING -> ACTIVE
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [ALERT] sofia.c:7724 sofia/internal/142 at 192.168.33.116:49383 receive message [ANSWER_EVENT]
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [ALERT] switch_core_state_machine.c:590 sofia/internal/142 at 192.168.33.116:49383 session thread sleep state: CS_CONSUME_MEDIA!
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [DEBUG] switch_core_codec.c:246 sofia/internal/1276 at domain.example.com Restore previous codec PCMU:0.
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [ALERT] switch_ivr_originate.c:3550 sofia/internal/1276 at domain.example.com receive message [ANSWER]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [DEBUG] mod_sofia.c:799 Local SDP sofia/internal/1276 at domain.example.com:
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 v=0
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 o=FreeSWITCH 1455357314 1455357316 IN IP4 219.206.20.22
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 s=FreeSWITCH
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 c=IN IP4 219.206.20.22
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 t=0 0
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 m=audio 26876 RTP/AVP 0 96
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=rtpmap:0 PCMU/8000
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=rtpmap:96 telephone-event/8000
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=fmtp:96 0-16
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=ptime:20
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=sendrecv
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [NOTICE] switch_ivr_originate.c:3550 Channel [sofia/internal/1276 at domain.example.com] has been answered
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [DEBUG] switch_channel.c:3767 (sofia/internal/1276 at domain.example.com) Callstate Change EARLY -> ACTIVE
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [ALERT] switch_ivr_originate.c:3550 sofia/internal/1276 at domain.example.com receive message [ANSWER_EVENT]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [DEBUG] sofia.c:6760 Channel sofia/internal/1276 at domain.example.com entering state [completed][200]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [DEBUG] switch_ivr_originate.c:3608 Originate Resulted in Success: [sofia/internal/142 at 192.168.33.116:49383]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [ALERT] switch_ivr_originate.c:3944 sofia/internal/1276 at domain.example.com receive message [AUDIO_SYNC]
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [ALERT] switch_core_state_machine.c:594 sofia/internal/142 at 192.168.33.116:49383 session thread wake state: CS_CONSUME_MEDIA!
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [ALERT] switch_ivr_originate.c:3941 sofia/internal/142 at 192.168.33.116:49383 receive message [AUDIO_SYNC]
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [ALERT] switch_core_state_machine.c:590 sofia/internal/142 at 192.168.33.116:49383 session thread sleep state: CS_CONSUME_MEDIA!
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [DEBUG] switch_ivr_originate.c:3608 Originate Resulted in Success: [sofia/internal/142 at 192.168.33.116:49383]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [ALERT] switch_ivr_originate.c:3944 sofia/internal/1276 at domain.example.com receive message [AUDIO_SYNC]
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [ALERT] switch_core_state_machine.c:594 sofia/internal/142 at 192.168.33.116:49383 session thread wake state: CS_CONSUME_MEDIA!
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [ALERT] switch_ivr_originate.c:3941 sofia/internal/142 at 192.168.33.116:49383 receive message [AUDIO_SYNC]
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [ALERT] switch_core_state_machine.c:590 sofia/internal/142 at 192.168.33.116:49383 session thread sleep state: CS_CONSUME_MEDIA!
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [ALERT] switch_core_state_machine.c:594 sofia/internal/142 at 192.168.33.116:49383 session thread wake state: CS_CONSUME_MEDIA!
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [ALERT] switch_ivr_bridge.c:1452 sofia/internal/142 at 192.168.33.116:49383 receive message [AUDIO_SYNC]
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [ALERT] switch_core_state_machine.c:590 sofia/internal/142 at 192.168.33.116:49383 session thread sleep state: CS_CONSUME_MEDIA!
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [ALERT] switch_ivr.c:217 sofia/internal/1276 at domain.example.com receive message [AUDIO_SYNC]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [ALERT] switch_ivr_bridge.c:1451 sofia/internal/1276 at domain.example.com receive message [AUDIO_SYNC]
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [ALERT] switch_ivr_bridge.c:1551 sofia/internal/142 at 192.168.33.116:49383 receive message [BRIDGE]
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [DEBUG] switch_core_media.c:9156 sofia/internal/142 at 192.168.33.116:49383 PAUSE Jitterbuffer
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [ALERT] switch_ivr_bridge.c:1559 sofia/internal/1276 at domain.example.com receive message [BRIDGE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [DEBUG] switch_core_media.c:9156 sofia/internal/1276 at domain.example.com PAUSE Jitterbuffer
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [DEBUG] switch_ivr_bridge.c:1591 (sofia/internal/142 at 192.168.33.116:49383) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [ALERT] switch_core_state_machine.c:594 sofia/internal/142 at 192.168.33.116:49383 session thread wake state: CS_CONSUME_MEDIA!
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at 192.168.33.116:49383) Running State Change CS_EXCHANGE_MEDIA
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [DEBUG] switch_core_state_machine.c:542 (sofia/internal/142 at 192.168.33.116:49383) State EXCHANGE_MEDIA
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [DEBUG] mod_sofia.c:613 SOFIA EXCHANGE_MEDIA
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.577825 [ALERT] switch_core_io.c:420 sofia/internal/1276 at domain.example.com receive message [TRANSCODING_NECESSARY]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.657897 [DEBUG] sofia.c:6760 Channel sofia/internal/1276 at domain.example.com entering state [ready][200]
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.667847 [ALERT] switch_ivr_bridge.c:283 sofia/internal/142 at 192.168.33.116:49383 receive message [DISPLAY]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.677907 [ALERT] switch_ivr_bridge.c:283 sofia/internal/1276 at domain.example.com receive message [DISPLAY]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.697815 [DEBUG] switch_rtp.c:6654 Correct audio ip/port confirmed.
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:15.367824 [INFO] switch_rtp.c:6616 Auto Changing audio port from 192.168.33.116:12552 to 73.170.102.43:12552
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:15.507826 [ALERT] switch_rtp.c:1612 sofia/internal/142 at 192.168.33.116:49383 Got: audio seq 17 but expected: 16 lost: 1
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:15.617825 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at domain.example.com Got: audio seq 57095 but expected: 57093 lost: 2
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:16.437897 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at domain.example.com Got: audio seq 57116 but expected: 57115 lost: 1
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:17.747957 [ALERT] switch_rtp.c:1612 sofia/internal/142 at 192.168.33.116:49383 Got: audio seq 129 but expected: 128 lost: 1
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:18.987891 [ALERT] switch_rtp.c:1612 sofia/internal/142 at 192.168.33.116:49383 Got: audio seq 191 but expected: 190 lost: 1
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:19.307915 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at domain.example.com Got: audio seq 57160 but expected: 57155 lost: 5
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:20.797895 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at domain.example.com Got: audio seq 57194 but expected: 57192 lost: 2
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:21.247947 [ALERT] switch_rtp.c:1612 sofia/internal/142 at 192.168.33.116:49383 Got: audio seq 304 but expected: 303 lost: 1
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:22.467916 [ALERT] switch_rtp.c:1612 sofia/internal/142 at 192.168.33.116:49383 Got: audio seq 365 but expected: 364 lost: 1
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:22.617927 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at domain.example.com Got: audio seq 57247 but expected: 57245 lost: 2
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:23.477916 [ALERT] switch_rtp.c:1612 sofia/internal/142 at 192.168.33.116:49383 Got: audio seq 416 but expected: 415 lost: 1
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:23.897894 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at domain.example.com Got: audio seq 57297 but expected: 57296 lost: 1
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:24.757895 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at domain.example.com Got: audio seq 57315 but expected: 57313 lost: 2
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:25.747894 [ALERT] switch_rtp.c:1612 sofia/internal/142 at 192.168.33.116:49383 Got: audio seq 529 but expected: 528 lost: 1
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/+12127773456 at 66.62.60.228:5060 [022e5d6e-83a2-4397-84ba-f0f49aa6eef6]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/+12127773456 at 66.62.60.228:5060) Running State Change CS_NEW
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] sofia.c:9248 sofia/internal/+12127773456 at 66.62.60.228:5060 receiving invite from 209.216.2.222:5660 version: 1.6.6  64bit
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_media.c:413 Looking for zrtp-hash
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_media.c:368 Deciding whether to pass zrtp-hash between legs
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_media.c:370 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
2016-02-13 09:23:25.807894 [DEBUG] sofia.c:9360 IP 209.216.2.222 Approved by acl "domains[]". Access Granted.
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] sofia.c:6760 Channel sofia/internal/+12127773456 at 66.62.60.228:5060 entering state [received][100]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] sofia.c:6770 Remote SDP:
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 v=0
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 o=Sonus_UAC 28164 6556 IN IP4 66.62.60.228
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 s=SIP Media Capabilities
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 c=IN IP4 66.62.60.229
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 t=0 0
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 m=audio 11570 RTP/AVP 18 0 101
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 a=rtpmap:18 G729/8000
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 a=fmtp:18 annexb=no
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 a=rtpmap:0 PCMU/8000
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 a=rtpmap:101 telephone-event/8000
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 a=fmtp:101 0-15
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 a=maxptime:20
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] sofia.c:7125 (sofia/internal/+12127773456 at 66.62.60.228:5060) State Change CS_NEW -> CS_INIT
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:492 (sofia/internal/+12127773456 at 66.62.60.228:5060) State NEW
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/+12127773456 at 66.62.60.228:5060) Running State Change CS_INIT
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/+12127773456 at 66.62.60.228:5060) State INIT
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] mod_sofia.c:88 sofia/internal/+12127773456 at 66.62.60.228:5060 SOFIA INIT
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:40 sofia/internal/+12127773456 at 66.62.60.228:5060 Standard INIT
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/+12127773456 at 66.62.60.228:5060) State Change CS_INIT -> CS_ROUTING
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/+12127773456 at 66.62.60.228:5060) State INIT going to sleep
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/+12127773456 at 66.62.60.228:5060) Running State Change CS_ROUTING
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_channel.c:2247 (sofia/internal/+12127773456 at 66.62.60.228:5060) Callstate Change DOWN -> RINGING
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/+12127773456 at 66.62.60.228:5060) State ROUTING
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] mod_sofia.c:141 sofia/internal/+12127773456 at 66.62.60.228:5060 SOFIA ROUTING
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:166 sofia/internal/+12127773456 at 66.62.60.228:5060 Standard ROUTING
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [INFO] mod_dialplan_xml.c:637 Processing FOOD BANK NYC <+12127773456>->19499300360 in context public
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 Dialplan: sofia/internal/+12127773456 at 66.62.60.228:5060 parsing [public->limit_exceeded] continue=false
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 Dialplan: sofia/internal/+12127773456 at 66.62.60.228:5060 Regex (FAIL) [limit_exceeded] destination_number(19499300360) =~ /^limit_exceeded$/ break=on-false
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 Dialplan: sofia/internal/+12127773456 at 66.62.60.228:5060 parsing [public->from_testopensip] continue=false
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 Dialplan: sofia/internal/+12127773456 at 66.62.60.228:5060 Regex (FAIL) [from_testopensip] network_addr(209.216.2.222) =~ /^209\.216\.15\.18$/ break=on-false
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 Dialplan: sofia/internal/+12127773456 at 66.62.60.228:5060 parsing [public->from_productionopensip] continue=false
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 Dialplan: sofia/internal/+12127773456 at 66.62.60.228:5060 Regex (PASS) [from_productionopensip] network_addr(209.216.2.222) =~ /^209\.216\.2\.222$/ break=on-false
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 Dialplan: sofia/internal/+12127773456 at 66.62.60.228:5060 Action set(dialed_domain=${sip_to_host})
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 Dialplan: sofia/internal/+12127773456 at 66.62.60.228:5060 Action set(domain_name=${sip_to_host})
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 Dialplan: sofia/internal/+12127773456 at 66.62.60.228:5060 Action set(company_name=${sip_to_host})
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 Dialplan: sofia/internal/+12127773456 at 66.62.60.228:5060 Action export(caller-id-in-from=true)
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 Dialplan: sofia/internal/+12127773456 at 66.62.60.228:5060 Action export(sip_cid_type=none)
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 Dialplan: sofia/internal/+12127773456 at 66.62.60.228:5060 Action lua(didhandle.lua ${destination_number})
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 Dialplan: sofia/internal/+12127773456 at 66.62.60.228:5060 Action log(INFO public Local URI var_name chan var is ${domain_name} Dialed domain is ${dialed_domain})
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:216 (sofia/internal/+12127773456 at 66.62.60.228:5060) State Change CS_ROUTING -> CS_EXECUTE
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/+12127773456 at 66.62.60.228:5060) State ROUTING going to sleep
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/+12127773456 at 66.62.60.228:5060) Running State Change CS_EXECUTE
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:539 (sofia/internal/+12127773456 at 66.62.60.228:5060) State EXECUTE
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] mod_sofia.c:196 sofia/internal/+12127773456 at 66.62.60.228:5060 SOFIA EXECUTE
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:258 sofia/internal/+12127773456 at 66.62.60.228:5060 Standard EXECUTE
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 set(dialed_domain=209.216.15.70)
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] mod_dptools.c:1498 SET sofia/internal/+12127773456 at 66.62.60.228:5060 [dialed_domain]=[209.216.15.70]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 set(domain_name=209.216.15.70)
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] mod_dptools.c:1498 SET sofia/internal/+12127773456 at 66.62.60.228:5060 [domain_name]=[209.216.15.70]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 set(company_name=209.216.15.70)
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] mod_dptools.c:1498 SET sofia/internal/+12127773456 at 66.62.60.228:5060 [company_name]=[209.216.15.70]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 export(caller-id-in-from=true)
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [caller-id-in-from]=[true]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 export(sip_cid_type=none)
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [sip_cid_type]=[none]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 lua(didhandle.lua 19499300360)
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC]
2016-02-13 09:23:25.807894 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f41ac03e390 Connected.
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 export(domain_name=209.216.15.70)
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [domain_name]=[209.216.15.70]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 export(skip_cdr_causes=LOSE_RACE)
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [skip_cdr_causes]=[LOSE_RACE]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [INFO] switch_cpp.cpp:1284 RPID is <sip:+12127773456 at 66.62.60.228:5060>;privacy=off dialing number is +12127773456
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 limit(hash +12127773456 0 1/10)
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_limit.c:126 incr called: +12127773456_0 max:1, interval:10
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] mod_hash.c:198 Usage for +12127773456_0 is now 1/1 for the last 10 seconds
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 limit(hash +12127773456 0 3/60)
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_limit.c:126 incr called: +12127773456_0 max:3, interval:60
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 set(effective_caller_id_name=FOOD BANK NYC)
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] mod_dptools.c:1498 SET sofia/internal/+12127773456 at 66.62.60.228:5060 [effective_caller_id_name]=[FOOD BANK NYC]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE]
2016-02-13 09:23:25.817894 [NOTICE] switch_cpp.cpp:1356 Product List Query SQL: select cebod_card.hold_music,cebod_card.status, cp.iduser, cp.product, cp.producttype as producttype, cp.activated, cd.domain, lower(cp.destinationtype) as destinationtype, cp.destination, cp.timecondition, cp.timeid, cp.timedestinationtype, cp.timedestination, cd.pbxip as pbxip, recording, whisper, whisper_file, postcall_id, substring(grp_description,1,15) as grpdescription  from cebod_products as cp, domain as cd left join cebod_card on cebod_card.id=iduser where  cp.iduser=cd.iduser and substr(cp.product,length(cp.product)-9,10) = substr('19499300360',length('19499300360')-9,10) 
2016-02-13 09:23:25.817894 [NOTICE] switch_cpp.cpp:1356 Time Availability : yes
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 set(api_hangup_hook=lua postcall.lua 1 domain.example.com)
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [DEBUG] mod_dptools.c:1498 SET sofia/internal/+12127773456 at 66.62.60.228:5060 [api_hangup_hook]=[lua postcall.lua 1 domain.example.com]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [INFO] switch_cpp.cpp:1284 Check id called id is blocked or not ,  notblockedSQL is select * from cebod_callblock where domain=domain.example.com and ( upper(phonenumber)=upper(substr(12127773456,1,length(phonenumber))) or upper(phonenumber)=upper(substr(FOOD BANK NYC,1,length(phonenumber))) ) limit 1
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [INFO] switch_cpp.cpp:1284 Check id called id+12127773456 is blocked or not ,  notblocked
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 export(calltype=origination)
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [calltype]=[origination]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 limit(hash +12127773456 17 1/10)
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [DEBUG] switch_limit.c:126 incr called: +12127773456_17 max:1, interval:10
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [DEBUG] mod_hash.c:198 Usage for +12127773456_17 is now 1/1 for the last 10 seconds
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 export(domain_name=domain.example.com)
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [domain_name]=[domain.example.com]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 export(accountcode=17)
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [accountcode]=[17]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 export(setinboundcalllimit=setlimitdomain.example.com 17 15)
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [setinboundcalllimit]=[setlimitdomain.example.com 17 15]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 limit(hash domain.example.com 17 15)
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [DEBUG] switch_limit.c:126 incr called: domain.example.com_17 max:15, interval:0
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [DEBUG] mod_hash.c:196 Usage for domain.example.com_17 is now 1/15
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.327895 [NOTICE] sofia.c:952 Hangup sofia/internal/1276 at domain.example.com [CS_EXECUTE] [NORMAL_CLEARING]
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_ivr_bridge.c:699 sofia/internal/1276 at domain.example.com ending bridge by request from write function
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_ivr_bridge.c:778 BRIDGE THREAD DONE [sofia/internal/142 at 192.168.33.116:49383]
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [NOTICE] switch_ivr_bridge.c:881 Hangup sofia/internal/142 at 192.168.33.116:49383 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_core_state_machine.c:542 (sofia/internal/142 at 192.168.33.116:49383) State EXCHANGE_MEDIA going to sleep
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at 192.168.33.116:49383) Running State Change CS_HANGUP
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_core_state_machine.c:739 (sofia/internal/142 at 192.168.33.116:49383) Callstate Change ACTIVE -> HANGUP
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/142 at 192.168.33.116:49383) State HANGUP
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] mod_sofia.c:425 sofia/internal/142 at 192.168.33.116:49383 Overriding SIP cause 480 with 200 from the other leg
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] mod_sofia.c:431 Channel sofia/internal/142 at 192.168.33.116:49383 hanging up, cause: NORMAL_CLEARING
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] mod_sofia.c:484 Sending BYE to sofia/internal/142 at 192.168.33.116:49383
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_core_state_machine.c:60 sofia/internal/142 at 192.168.33.116:49383 Standard HANGUP, cause: NORMAL_CLEARING
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/142 at 192.168.33.116:49383) State HANGUP going to sleep
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_core_state_machine.c:508 (sofia/internal/142 at 192.168.33.116:49383) State Change CS_HANGUP -> CS_REPORTING
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at 192.168.33.116:49383) Running State Change CS_REPORTING
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/142 at 192.168.33.116:49383) State REPORTING
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_core_state_machine.c:104 sofia/internal/142 at 192.168.33.116:49383 Standard REPORTING, cause: NORMAL_CLEARING
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/142 at 192.168.33.116:49383) State REPORTING going to sleep
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/142 at 192.168.33.116:49383) State Change CS_REPORTING -> CS_DESTROY
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_core_session.c:1646 Session 1857 (sofia/internal/142 at 192.168.33.116:49383) Locked, Waiting on external entities
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [DEBUG] switch_ivr_bridge.c:705 sofia/internal/1276 at domain.example.com ending bridge by request from read function
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [DEBUG] switch_ivr_bridge.c:778 BRIDGE THREAD DONE [sofia/internal/1276 at domain.example.com]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [ALERT] switch_ivr_bridge.c:1692 sofia/internal/1276 at domain.example.com receive message [UNBRIDGE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [DEBUG] switch_ivr_bridge.c:1692 sofia/internal/1276 at domain.example.com skip receive message [UNBRIDGE] (channel is hungup already)
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.337895 [NOTICE] switch_core_session.c:1664 Session 1857 (sofia/internal/142 at 192.168.33.116:49383) Ended
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.337895 [NOTICE] switch_core_session.c:1668 Close Channel sofia/internal/142 at 192.168.33.116:49383 [CS_DESTROY]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [DEBUG] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [DEBUG] switch_core_state_machine.c:539 (sofia/internal/1276 at domain.example.com) State EXECUTE going to sleep
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at domain.example.com) Running State Change CS_HANGUP
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.337895 [DEBUG] switch_core_state_machine.c:630 (sofia/internal/142 at 192.168.33.116:49383) Running State Change CS_DESTROY
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.337895 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/142 at 192.168.33.116:49383) State DESTROY
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.337895 [DEBUG] mod_sofia.c:341 sofia/internal/142 at 192.168.33.116:49383 SOFIA DESTROY
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.337895 [DEBUG] switch_core_state_machine.c:111 sofia/internal/142 at 192.168.33.116:49383 Standard DESTROY
d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.337895 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/142 at 192.168.33.116:49383) State DESTROY going to sleep
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [DEBUG] switch_core_state_machine.c:739 (sofia/internal/1276 at domain.example.com) Callstate Change ACTIVE -> HANGUP
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/1276 at domain.example.com) State HANGUP
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [DEBUG] mod_sofia.c:431 Channel sofia/internal/1276 at domain.example.com hanging up, cause: NORMAL_CLEARING
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1276 at domain.example.com Standard HANGUP, cause: NORMAL_CLEARING
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/1276 at domain.example.com) State HANGUP going to sleep
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [DEBUG] switch_core_state_machine.c:508 (sofia/internal/1276 at domain.example.com) State Change CS_HANGUP -> CS_REPORTING
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at domain.example.com) Running State Change CS_REPORTING
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/1276 at domain.example.com) State REPORTING
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [DEBUG] mod_odbc_cdr.c:309 sql INSERT INTO cebod_cdr (used_gateway_id, phonenumber, call_uuid, leg, job_uuid, callrequest_id, accountcode, created_date, callerid, billsec, callername, starting_date, hangup_cause, duration, event_name) VALUES (68.5.94.63, 142, a17a5d15-d97e-4c70-b476-bd0ff4ac2e09, inbound, a17a5d15-d97e-4c70-b476-bd0ff4ac2e09, a17a5d15-d97e-4c70-b476-bd0ff4ac2e09, 17, 2016-02-13 09:23:26, 1276, 12, Kunal Mittal, 2016-02-13 09:23:10, NORMAL_CLEARING, 16, REQUEST_PARAMS)
2016-02-13 09:23:26.337895 [ALERT] switch_odbc.c:353 Connecting opensips
2016-02-13 09:23:26.337895 [ALERT] switch_odbc.c:397 Connected to [opensips]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.347896 [DEBUG] switch_core_state_machine.c:104 sofia/internal/1276 at domain.example.com Standard REPORTING, cause: NORMAL_CLEARING
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.347896 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/1276 at domain.example.com) State REPORTING going to sleep
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.347896 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/1276 at domain.example.com) State Change CS_REPORTING -> CS_DESTROY
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.347896 [DEBUG] switch_core_session.c:1646 Session 1856 (sofia/internal/1276 at domain.example.com) Locked, Waiting on external entities
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.347896 [NOTICE] switch_core_session.c:1664 Session 1856 (sofia/internal/1276 at domain.example.com) Ended
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.347896 [NOTICE] switch_core_session.c:1668 Close Channel sofia/internal/1276 at domain.example.com [CS_DESTROY]
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.347896 [DEBUG] switch_core_state_machine.c:630 (sofia/internal/1276 at domain.example.com) Running State Change CS_DESTROY
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.347896 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/1276 at domain.example.com) State DESTROY
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.347896 [DEBUG] mod_sofia.c:341 sofia/internal/1276 at domain.example.com SOFIA DESTROY
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.347896 [DEBUG] switch_core_state_machine.c:111 sofia/internal/1276 at domain.example.com Standard DESTROY
a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.347896 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/1276 at domain.example.com) State DESTROY going to sleep
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [INFO] switch_cpp.cpp:1284 Domain name is : 219.206.20.22dialed number 19499300360 destiantiotype is ivr destination is 36 account activation status: 1
2016-02-13 09:23:26.827933 [NOTICE] switch_cpp.cpp:1356 Get Special Rate inforamtion: select billingblock, per_min_did as rate from cebod_specialrate where id_cc_card in (0,17) order by id_cc_card desc limit 1
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [INFO] switch_cpp.cpp:1284 Sending Call to  IVR, IVR ID is 36
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 lua(ivr_menu.lua 36 domain.example.com)
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC]
2016-02-13 09:23:26.827933 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f4164033540 Connected.
2016-02-13 09:23:26.827933 [NOTICE] switch_cpp.cpp:1356 [ivr_menu] SQL: SELECT * FROM cebod_ivr_menus
                WHERE ivr_id = '36'
                AND ivr_menu_enabled = 'true' 
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [ALERT] switch_cpp.cpp:683 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [ANSWER]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [ALERT] switch_core_media.c:413 Looking for zrtp-hash
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [ALERT] switch_core_media.c:368 Deciding whether to pass zrtp-hash between legs
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [ALERT] switch_core_media.c:373 No partner channel found, so not propagating zrtp-hash
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [G729:18:8000:20:8000:1]/[PCMU:0:8000:20:64000:1]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [G729:18:8000:20:8000:1]/[PCMA:8:8000:20:64000:1]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [G729:18:8000:20:8000:1]/[G729:18:8000:20:8000:1]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [G729:18:8000:20:8000:1] ++++ is saved as a match
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [G729:18:8000:20:8000:1]/[GSM:3:8000:20:13200:1]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G729:18:8000:20:8000:1]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:4077 Set telephone-event payload to 101 at 8000
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:2906 Set Codec sofia/internal/+12127773456 at 66.62.60.228:5060 G729/8000 20 ms 160 samples 8000 bits 1 channels
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_codec.c:111 sofia/internal/+12127773456 at 66.62.60.228:5060 Original read codec set to G729:18
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:4429 Set telephone-event payload to 101 at 8000
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:4485 sofia/internal/+12127773456 at 66.62.60.228:5060 Set 2833 dtmf send payload to 101 recv payload to 101
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:6033 AUDIO RTP [sofia/internal/+12127773456 at 66.62.60.228:5060] 219.206.20.22 port 28274 -> 66.62.60.229 port 11570 codec: 18 ms: 20
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_rtp.c:3802 Starting timer [soft] 160 bytes per 20ms
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:1939 Setting Jitterbuffer to 60ms (3 frames) (50 max frames)
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:6332 sofia/internal/+12127773456 at 66.62.60.228:5060 Set 2833 dtmf send payload to 101
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:6339 sofia/internal/+12127773456 at 66.62.60.228:5060 Set 2833 dtmf receive payload to 101
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:6362 sofia/internal/+12127773456 at 66.62.60.228:5060 Set rtp dtmf delay to 60
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/+12127773456 at 66.62.60.228:5060!
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_channel.c:3468 (sofia/internal/+12127773456 at 66.62.60.228:5060) Callstate Change RINGING -> EARLY
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [ALERT] sofia_media.c:92 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [PROGRESS_EVENT]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] mod_sofia.c:799 Local SDP sofia/internal/+12127773456 at 66.62.60.228:5060:
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 v=0
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 o=FreeSWITCH 1455355932 1455355933 IN IP4 219.206.20.22
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 s=FreeSWITCH
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 c=IN IP4 219.206.20.22
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 t=0 0
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 m=audio 28274 RTP/AVP 18 101
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 a=rtpmap:18 G729/8000
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 a=fmtp:18 annexb=no
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 a=rtpmap:101 telephone-event/8000
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 a=fmtp:101 0-16
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 a=ptime:20
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 a=sendrecv
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [NOTICE] switch_cpp.cpp:683 Channel [sofia/internal/+12127773456 at 66.62.60.228:5060] has been answered
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_channel.c:3767 (sofia/internal/+12127773456 at 66.62.60.228:5060) Callstate Change EARLY -> ACTIVE
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [ALERT] switch_cpp.cpp:683 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [ANSWER_EVENT]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] sofia.c:6760 Channel sofia/internal/+12127773456 at 66.62.60.228:5060 entering state [completed][200]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.887900 [DEBUG] sofia.c:6760 Channel sofia/internal/+12127773456 at 66.62.60.228:5060 entering state [ready][200]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.957827 [DEBUG] switch_rtp.c:6654 Correct audio ip/port confirmed.

022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:27.837896 [DEBUG] switch_ivr_play_say.c:1359 Codec Activated L16 at 8000hz 1 channels 20ms
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:27.837896 [ALERT] switch_ivr_play_say.c:1290 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [AUDIO_SYNC]
022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:27.857913 [ALERT] switch_core_io.c:1135 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [TRANSCODING_NECESSARY]
2016-02-13 09:23:27.857913 [INFO] mod_com_g729.c:126 ENCODER LICENSE ALLOCATED--->0x7f418400f440 0x7f418400f440
2016-02-13 09:23:27.857913 [INFO] mod_com_g729.c:133 ENCODER CREATED------------->0x7f418400f440 0x7f418400f440
2016-02-13 09:23:29.137959 [ALERT] switch_odbc.c:353 Connecting opensips
2016-02-13 09:23:29.137959 [ALERT] switch_odbc.c:397 Connected to [opensips]
2016-02-13 09:23:29.137959 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f424c24f000 Connected.
2016-02-13 09:23:29.157902 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f424c24f000 released.
2016-02-13 09:23:32.567895 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f424c24f000 Connected.
2016-02-13 09:23:32.567895 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f424c24f000 released.
2016-02-13 09:23:34.197900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/124.csv
2016-02-13 09:23:34.197900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/164.csv
2016-02-13 09:23:34.197900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/638.csv
2016-02-13 09:23:34.197900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/474.csv
2016-02-13 09:23:34.197900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/290.csv
2016-02-13 09:23:34.197900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/128.csv
2016-02-13 09:23:34.197900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/608.csv
2016-02-13 09:23:34.197900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/428.csv
2016-02-13 09:23:34.197900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/374.csv
2016-02-13 09:23:34.197900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/17.csv
2016-02-13 09:23:34.197900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/236.csv
2016-02-13 09:23:34.197900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/322.csv
2016-02-13 09:23:34.197900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/432.csv
-------------- next part --------------
2016-02-13 09:21:50.767867 [NOTICE] mod_logfile.c:213 New log started.
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.607889 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/142 at domain.example.com:5660 [03d4582f-ed06-4268-a70b-0dd9307fb4cf]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.607889 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at domain.example.com:5660) Running State Change CS_NEW
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.607889 [DEBUG] sofia.c:9248 sofia/internal/142 at domain.example.com:5660 receiving invite from 73.170.102.43:49383 version: 1.6.6  64bit
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.607889 [ALERT] switch_core_media.c:413 Looking for zrtp-hash
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.607889 [ALERT] switch_core_media.c:368 Deciding whether to pass zrtp-hash between legs
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.607889 [ALERT] switch_core_media.c:370 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
2016-02-13 09:21:55.607889 [DEBUG] sofia.c:9415 IP 73.170.102.43 Rejected by acl "domains". Falling back to Digest auth.
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.607889 [DEBUG] switch_core_state_machine.c:492 (sofia/internal/142 at domain.example.com:5660) State NEW
2016-02-13 09:21:55.607889 [DEBUG] sofia.c:2147 detaching session 03d4582f-ed06-4268-a70b-0dd9307fb4cf
2016-02-13 09:21:55.637867 [DEBUG] sofia.c:2255 Re-attaching to session 03d4582f-ed06-4268-a70b-0dd9307fb4cf
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.647889 [DEBUG] sofia.c:9248 sofia/internal/142 at domain.example.com:5660 receiving invite from 73.170.102.43:49383 version: 1.6.6  64bit
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.647889 [ALERT] switch_core_media.c:413 Looking for zrtp-hash
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.647889 [ALERT] switch_core_media.c:368 Deciding whether to pass zrtp-hash between legs
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.647889 [ALERT] switch_core_media.c:370 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
2016-02-13 09:21:55.647889 [DEBUG] sofia.c:9415 IP 73.170.102.43 Rejected by acl "domains". Falling back to Digest auth.
2016-02-13 09:21:55.647889 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f41ac03e390 Connected.
2016-02-13 09:21:55.657791 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f41ac03e390 released.
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] sofia.c:10549 Setting NAT mode based on nat.auto
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] sofia.c:6760 Channel sofia/internal/142 at domain.example.com:5660 entering state [received][100]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] sofia.c:6770 Remote SDP:
03d4582f-ed06-4268-a70b-0dd9307fb4cf v=0
03d4582f-ed06-4268-a70b-0dd9307fb4cf o=142 8000 8000 IN IP4 192.168.33.116
03d4582f-ed06-4268-a70b-0dd9307fb4cf s=SIP Call
03d4582f-ed06-4268-a70b-0dd9307fb4cf c=IN IP4 192.168.33.116
03d4582f-ed06-4268-a70b-0dd9307fb4cf t=0 0
03d4582f-ed06-4268-a70b-0dd9307fb4cf m=audio 50028 RTP/AVP 0 8 101
03d4582f-ed06-4268-a70b-0dd9307fb4cf a=rtpmap:0 PCMU/8000
03d4582f-ed06-4268-a70b-0dd9307fb4cf a=rtpmap:8 PCMA/8000
03d4582f-ed06-4268-a70b-0dd9307fb4cf a=rtpmap:101 telephone-event/8000
03d4582f-ed06-4268-a70b-0dd9307fb4cf a=fmtp:101 0-15
03d4582f-ed06-4268-a70b-0dd9307fb4cf a=ptime:20
03d4582f-ed06-4268-a70b-0dd9307fb4cf 
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] sofia.c:7125 (sofia/internal/142 at domain.example.com:5660) State Change CS_NEW -> CS_INIT
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at domain.example.com:5660) Running State Change CS_INIT
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/142 at domain.example.com:5660) State INIT
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] mod_sofia.c:88 sofia/internal/142 at domain.example.com:5660 SOFIA INIT
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:40 sofia/internal/142 at domain.example.com:5660 Standard INIT
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/142 at domain.example.com:5660) State Change CS_INIT -> CS_ROUTING
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/142 at domain.example.com:5660) State INIT going to sleep
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at domain.example.com:5660) Running State Change CS_ROUTING
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_channel.c:2247 (sofia/internal/142 at domain.example.com:5660) Callstate Change DOWN -> RINGING
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/142 at domain.example.com:5660) State ROUTING
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] mod_sofia.c:141 sofia/internal/142 at domain.example.com:5660 SOFIA ROUTING
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:166 sofia/internal/142 at domain.example.com:5660 Standard ROUTING
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [INFO] mod_dialplan_xml.c:637 Processing 142 <142>->1276 in context public
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [public->limit_exceeded] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [limit_exceeded] destination_number(1276) =~ /^limit_exceeded$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [public->from_testopensip] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [from_testopensip] network_addr(73.170.102.43) =~ /^209\.216\.15\.18$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [public->from_productionopensip] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [from_productionopensip] network_addr(73.170.102.43) =~ /^209\.216\.2\.222$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [public->from_registeredusers] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Absolute Condition [from_registeredusers]
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action set(dialed_domain=${sip_to_host})
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action set(domain_name=${sip_to_host})
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action set(company_name=${sip_to_host})
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action export(caller-id-in-from=true)
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action export(sip_cid_type=none)
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action log(Public IP is ${bind_server_ip} Domain is ${domain_name} Dialed domain is ${dialed_domain})
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action lua(didhandle.lua ${destination_number})
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action log(INFO public Local URI var_name chan var is ${domain_name} Dialed domain is ${dialed_domain})
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:216 (sofia/internal/142 at domain.example.com:5660) State Change CS_ROUTING -> CS_EXECUTE
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/142 at domain.example.com:5660) State ROUTING going to sleep
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at domain.example.com:5660) Running State Change CS_EXECUTE
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:539 (sofia/internal/142 at domain.example.com:5660) State EXECUTE
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] mod_sofia.c:196 sofia/internal/142 at domain.example.com:5660 SOFIA EXECUTE
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:258 sofia/internal/142 at domain.example.com:5660 Standard EXECUTE
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(dialed_domain=domain.example.com)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [dialed_domain]=[domain.example.com]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(domain_name=domain.example.com)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [domain_name]=[domain.example.com]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(company_name=domain.example.com)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [company_name]=[domain.example.com]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 export(caller-id-in-from=true)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [caller-id-in-from]=[true]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 export(sip_cid_type=none)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [sip_cid_type]=[none]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 log(Public IP is 219.206.20.22 Domain is domain.example.com Dialed domain is domain.example.com)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] mod_dptools.c:1692 IP is 219.206.20.22 Domain is domain.example.com Dialed domain is domain.example.com
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 lua(didhandle.lua 1276)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
2016-02-13 09:21:55.657791 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f41ac03e390 Connected.
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 export(domain_name=domain.example.com)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [domain_name]=[domain.example.com]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 export(skip_cdr_causes=LOSE_RACE)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [skip_cdr_causes]=[LOSE_RACE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [INFO] switch_cpp.cpp:1284 RPID is  dialing number is 142
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 export(accountcode=17)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [accountcode]=[17]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [INFO] switch_cpp.cpp:1284 String length is less that 7 should be extension
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(process_cdr=a_only)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [process_cdr]=[a_only]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_ivr.c:2085 (sofia/internal/142 at domain.example.com:5660) State Change CS_EXECUTE -> CS_ROUTING
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_ivr.c:2090 sofia/internal/142 at domain.example.com:5660 receive message [TRANSFER]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [NOTICE] switch_ivr.c:2092 Transfer sofia/internal/142 at domain.example.com:5660 to XML[1276 at default]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_cpp.cpp:898 transfer result: 0
2016-02-13 09:21:55.657791 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f41ac03e390 released.
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_cpp.cpp:1103 sofia/internal/142 at domain.example.com:5660 destroy/unlink session from object
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:539 (sofia/internal/142 at domain.example.com:5660) State EXECUTE going to sleep
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at domain.example.com:5660) Running State Change CS_ROUTING
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/142 at domain.example.com:5660) State ROUTING
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] mod_sofia.c:141 sofia/internal/142 at domain.example.com:5660 SOFIA ROUTING
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:166 sofia/internal/142 at domain.example.com:5660 Standard ROUTING
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [INFO] mod_dialplan_xml.c:637 Processing 142 <142>->1276 in context default
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->longdistance] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [longdistance] destination_number(1276) =~ /^\d{10,15}$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->unloop] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->freeswitch_public_conf_via_sip] continue=false
2016-02-13 09:21:55.657791 [ERR] switch_regex.c:104 COMPILE ERROR: 1 [nothing to repeat][^*9(888|8888|1616|3232)$]
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [freeswitch_public_conf_via_sip] destination_number(1276) =~ /^*9(888|8888|1616|3232)$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->tod_example] continue=true
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Date/TimeMatch (FAIL) [tod_example] break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->holiday_example] continue=true
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Date/TimeMatch (FAIL) [holiday_example] break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->global-intercept] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [global-intercept] destination_number(1276) =~ /^\*886$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->group-intercept] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [group-intercept] destination_number(1276) =~ /^\*8$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->intercept-ext] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [intercept-ext] destination_number(1276) =~ /^\*\*(\d+)$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->redial] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [redial] destination_number(1276) =~ /^\*(redial|870)$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->global] continue=true
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (PASS) [global] ${endpoint_disposition}(DELAYED NEGOTIATION) =~ /^(DELAYED NEGOTIATION)/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [global] ${switch_r_sdp}(v=0
03d4582f-ed06-4268-a70b-0dd9307fb4cf o=142 8000 8000 IN IP4 192.168.33.116
03d4582f-ed06-4268-a70b-0dd9307fb4cf s=SIP Call
03d4582f-ed06-4268-a70b-0dd9307fb4cf c=IN IP4 192.168.33.116
03d4582f-ed06-4268-a70b-0dd9307fb4cf t=0 0
03d4582f-ed06-4268-a70b-0dd9307fb4cf m=audio 50028 RTP/AVP 0 8 101
03d4582f-ed06-4268-a70b-0dd9307fb4cf a=rtpmap:0 PCMU/8000
03d4582f-ed06-4268-a70b-0dd9307fb4cf a=rtpmap:8 PCMA/8000
03d4582f-ed06-4268-a70b-0dd9307fb4cf a=rtpmap:101 telephone-event/8000
03d4582f-ed06-4268-a70b-0dd9307fb4cf a=fmtp:101 0-15
03d4582f-ed06-4268-a70b-0dd9307fb4cf a=ptime:20
03d4582f-ed06-4268-a70b-0dd9307fb4cf ) =~ /(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)/ break=never
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Absolute Condition [global]
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action hash(insert/${domain_name}-last_dial/global/${uuid})
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->snom-demo-2] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [snom-demo-2] destination_number(1276) =~ /^9001$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->snom-demo-1] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [snom-demo-1] destination_number(1276) =~ /^9000$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->eavesdrop] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [eavesdrop] destination_number(1276) =~ /^\*88(\d{3,5})$|^\*0(.*)$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->eavesdrop] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [eavesdrop] destination_number(1276) =~ /^\*779$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->call_return] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [call_return] destination_number(1276) =~ /^\*69$|^\*869$|^lcr$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->del-number] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [del-number] destination_number(1276) =~ /^\*60$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->add-number] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [add-number] destination_number(1276) =~ /^\*61$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->check-number] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [check-number] destination_number(1276) =~ /^\*62$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->enable-time-condition] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [enable-time-condition] destination_number(1276) =~ /^\*63$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->disable-time-condition] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [disable-time-condition] destination_number(1276) =~ /^\*64$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->del-group] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [del-group] destination_number(1276) =~ /^\*80$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->add-group] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [add-group] destination_number(1276) =~ /^\*81$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->call-group-simo] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [call-group-simo] destination_number(1276) =~ /^\*82(\d{2})$|^82(\d{2})$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->call-group-order] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [call-group-order] destination_number(1276) =~ /^\*83(\d{2})$|^83(\d{2})$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->extension-intercom] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [extension-intercom] destination_number(1276) =~ /^\*84(\d{3,5})$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->user_recodring_enabled] continue=true
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (PASS) [user_recodring_enabled] ${recording}(off) =~ /^(on|off)$/ break=on-true
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->user_callscreening] continue=true
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [user_callscreening] caller_id_number(142) =~ /^\d{10,15}$/ break=on-true
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (PASS) [user_callscreening] ${callscreening}(off) =~ /^(on|off)$/ break=on-true
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->Check IVR-based CF] continue=true
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (PASS) [Check IVR-based CF] destination_number(1276) =~ /^(\d+)$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action set(dialed_number=1276) INLINE
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(dialed_number=1276)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [dialed_number]=[1276]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action set(cf_target=${db(select/${domain_name}-CF/1276)}) INLINE
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(cf_target=)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [cf_target]=[UNDEF]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [Check IVR-based CF] ${recording}(off) =~ /^on$/ break=never
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [Check IVR-based CF] ${cf_target}() =~ /^\d{3,5}$/ break=never
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [Check IVR-based CF] ${cf_target}() =~ /^\d{10,15}$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->CF from my station] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [CF from my station] destination_number(1276) =~ /^\*72$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->CF cancel from my station] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [CF cancel from my station] destination_number(1276) =~ /^\*73$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->CF from any station] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [CF from any station] destination_number(1276) =~ /^\*76$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->CF cancel from any station] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [CF cancel from any station] destination_number(1276) =~ /^\*77$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->NACF from my station] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [NACF from my station] destination_number(1276) =~ /^\*74$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->NACF cancel from my station] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [NACF cancel from my station] destination_number(1276) =~ /^\*75$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->Add_Member_To_Queue] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [Add_Member_To_Queue] destination_number(1276) =~ /^\*51$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->Delete_Member_From_Queue] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [Delete_Member_From_Queue] destination_number(1276) =~ /^\*52$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->Pickup_Call_From_Queue] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [Pickup_Call_From_Queue] destination_number(1276) =~ /^\*53$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->dial-by-name] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [dial-by-name] destination_number(1276) =~ /^\*40$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->conference_withpin] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [conference_withpin] destination_number(1276) =~ /^\*3560$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->Voicemail_check] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [Voicemail_check] caller_id_number(142) =~ /^(1276)$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->check_availability] continue=true
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [check_availability] destination_number(1276) =~ /^hkjhk(1[0-9][0-9][0-9]|[1-2][0-9][0-9])$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->is_check_availability] continue=true
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [is_check_availability] available_to_call() =~ /true/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->osbridge] continue=true
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (PASS) [osbridge] destination_number(1276) =~ /(^\d{3,5})$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action set(proxy_media=false)
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action set(continue_on_fail=true)
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action export(dialed_extension=1276) INLINE
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 export(dialed_extension=1276)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [dialed_extension]=[1276]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action export(domain_name=${domain_name})
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action set(accountcode=${user_data(${dialed_extension}@${domain_name} var accountcode)})
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action export(available_to_call=true) INLINE
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 export(available_to_call=true)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [available_to_call]=[true]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action log(Extensions ${dialed_extension} is ${available_to_call})
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action set(nacf_target=${db(select/${domain_name}-NACF/1276)}) INLINE
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(nacf_target=)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [nacf_target]=[UNDEF]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [osbridge] ${recording}(off) =~ /^on$/ break=never
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (PASS) [osbridge] ${available_to_call}(true) =~ /^true$/ break=never
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action bind_meta_app(1 b s execute_extension::dx XML features)
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/var/lib/freeswitch/recordings/${domain_name}/${strftime(%Y-%m-%d-%H-%M-%S)}.${uuid}.wav)
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action bind_meta_app(3 b s execute_extension::cf XML features)
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action bind_meta_app(4 b s execute_extension::attented_xfer XML default)
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action bind_meta_app(5 b s execute_extension::attented_xfer_to_vm XML default)
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action bind_meta_app(6 b s execute_extension::blind_xfer_to XML default)
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action bind_meta_app(9 b s execute_extension::call_park XML default)
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action export(sip_contact_user=${sip_from_user})
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action set(ringback=${us-ring})
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action set(transfer_ringback=local_stream://moh)
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action set(call_timeout=35)
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action set(hangup_after_bridge=true)
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action bridge(user/${dialed_extension}@${domain_name})
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action sleep(1000)
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [osbridge] ${nacf_target}() =~ /^\d{3,7}$/ break=never
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [osbridge] ${nacf_target}() =~ /^\d{10,15}$/ break=never
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Absolute Condition [osbridge]
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action answer()
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action sleep(500)
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action voicemail(default ${domain_name} ${dialed_extension})
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->Local_Extension_VM] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [Local_Extension_VM] destination_number(1276) =~ /^vm-(\d{3,5})$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->attented_xfer_to_vm] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [attented_xfer_to_vm] destination_number(1276) =~ /^attented_xfer_to_vm$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->attented_xfer] continue=true
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [attented_xfer] destination_number(1276) =~ /^attented_xfer$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->blind_xfer_to] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [blind_xfer_to] destination_number(1276) =~ /^blind_xfer_to$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->Local_Extension] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [Local_Extension] destination_number(1276) =~ /^disable-(1[0-9][0-9][0-9]|[1-2][0-9][0-9])$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->Local_Extension_Skinny] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [Local_Extension_Skinny] destination_number(1276) =~ /^(11[01][0-9])$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->park-in] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [park-in] destination_number(1276) =~ /^call_park$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->park-out] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [park-out] destination_number(1276) =~ /^\*(85\d\d)$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->vmain] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [vmain] destination_number(1276) =~ /^vmain$|^\*98$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->sip_uri] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [sip_uri] destination_number(1276) =~ /^sip:(.*)$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->local_uri] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [local_uri] destination_number(1276) =~ /^(local_.*)$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->nb_conferences] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [nb_conferences] destination_number(1276) =~ /^\*(30\d{2,4})$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->wb_conferences] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [wb_conferences] destination_number(1276) =~ /^\*(31\d{2})$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->uwb_conferences] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [uwb_conferences] destination_number(1276) =~ /^\*(32\d{2})$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->cdquality_conferences] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [cdquality_conferences] destination_number(1276) =~ /^\*(33\d{2})$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->mad_boss_intercom] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [mad_boss_intercom] destination_number(1276) =~ /^0911$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->mad_boss_intercom] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [mad_boss_intercom] destination_number(1276) =~ /^0912$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->mad_boss] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [mad_boss] destination_number(1276) =~ /^0913$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->queupark] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [queupark] destination_number(1276) =~ /^\*(5902)$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->queueunpark] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [queueunpark] destination_number(1276) =~ /^\*(5903)$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->park] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [park] destination_number(1276) =~ /^\*5900$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->unpark] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [unpark] destination_number(1276) =~ /^\*(5901)$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->valet_park] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [valet_park] destination_number(1276) =~ /^(6000)$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->valet_park] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [valet_park] destination_number(1276) =~ /^(60\d[1-9])$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->park] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [park] destination_number(1276) =~ /park\+(\d+)/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->unpark] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [unpark] destination_number(1276) =~ /^parking$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->park] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [park] destination_number(1276) =~ /callpark/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->unpark] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [unpark] destination_number(1276) =~ /pickup/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->wait] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [wait] destination_number(1276) =~ /^wait$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->fax_receive] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [fax_receive] destination_number(1276) =~ /^fax.*$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->fax_transmit] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [fax_transmit] destination_number(1276) =~ /^9179$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->ringback_180] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [ringback_180] destination_number(1276) =~ /^9180$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->ringback_183_uk_ring] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [ringback_183_uk_ring] destination_number(1276) =~ /^\*9181$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->ringback_183_music_ring] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [ringback_183_music_ring] destination_number(1276) =~ /^\*9182$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->ringback_post_answer_uk_ring] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [ringback_post_answer_uk_ring] destination_number(1276) =~ /^\*9183$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->ringback_post_answer_music] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [ringback_post_answer_music] destination_number(1276) =~ /^\*9184$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->show_info] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [show_info] destination_number(1276) =~ /^\*9192$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->video_record] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [video_record] destination_number(1276) =~ /^\*9193$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->video_playback] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [video_playback] destination_number(1276) =~ /^\*9194$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->delay_echo] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [delay_echo] destination_number(1276) =~ /^\*9195$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->echo] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [echo] destination_number(1276) =~ /^\*9196$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->milliwatt] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [milliwatt] destination_number(1276) =~ /^\*9197$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->tone_stream] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [tone_stream] destination_number(1276) =~ /^\*9198$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->zrtp_enrollement] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [zrtp_enrollement] destination_number(1276) =~ /^\*9787$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->hold_music] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [hold_music] destination_number(1276) =~ /^\*9664$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->laugh break] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [laugh break] destination_number(1276) =~ /^\*9386$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->101] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [101] destination_number(1276) =~ /^101$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->pizza_demo] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [pizza_demo] destination_number(1276) =~ /^(pizza|74992)$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->Talking Clock Time] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [Talking Clock Time] destination_number(1276) =~ /^9170$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->Talking Clock Date] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [Talking Clock Date] destination_number(1276) =~ /^9171$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->Talking Clock Date and Time] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [Talking Clock Date and Time] destination_number(1276) =~ /^9172$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->local.example.com] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [local.example.com] ${toll_allow}() =~ /local/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->domestic.example.com] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [domestic.example.com] ${toll_allow}() =~ /domestic/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->international.example.com] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [international.example.com] ${toll_allow}() =~ /international/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->conference] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [conference] destination_number(1276) =~ /^112233$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->didforsale_ivr] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [didforsale_ivr] destination_number(1276) =~ /5566/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->domian_ivr] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [domian_ivr] destination_number(1276) =~ /ivr(1949.*)$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->enum] continue=false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (PASS) [enum] ${module_exists(mod_enum)}(true) =~ /true/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (PASS) [enum] destination_number(1276) =~ /^(.*)$/ break=on-false
03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action transfer(1276 enum)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] switch_core_state_machine.c:216 (sofia/internal/142 at domain.example.com:5660) State Change CS_ROUTING -> CS_EXECUTE
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/142 at domain.example.com:5660) State ROUTING going to sleep
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at domain.example.com:5660) Running State Change CS_EXECUTE
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] switch_core_state_machine.c:539 (sofia/internal/142 at domain.example.com:5660) State EXECUTE
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] mod_sofia.c:196 sofia/internal/142 at domain.example.com:5660 SOFIA EXECUTE
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] switch_core_state_machine.c:258 sofia/internal/142 at domain.example.com:5660 Standard EXECUTE
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 hash(insert/domain.example.com-spymap/142/03d4582f-ed06-4268-a70b-0dd9307fb4cf)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 hash(insert/domain.example.com-last_dial/142/1276)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 hash(insert/domain.example.com-last_dial/global/03d4582f-ed06-4268-a70b-0dd9307fb4cf)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 export(RFC2822_DATE=Sat, 13 Feb 2016 09:21:55 -0800)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [RFC2822_DATE]=[Sat, 13 Feb 2016 09:21:55 -0800]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(proxy_media=false)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [proxy_media]=[false]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(continue_on_fail=true)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [continue_on_fail]=[true]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 export(domain_name=domain.example.com)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [domain_name]=[domain.example.com]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
2016-02-13 09:21:55.667790 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f41ac03e390 Connected.
2016-02-13 09:21:55.667790 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f41ac03e390 released.
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(accountcode=17)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [accountcode]=[17]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 log(Extensions 1276 is true)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] mod_dptools.c:1692 1276 is true
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 bind_meta_app(1 b s execute_extension::dx XML features)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *1 execute_extension::dx XML features
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 bind_meta_app(2 b s record_session::/usr/local/freeswitch/var/lib/freeswitch/recordings/domain.example.com/2016-02-13-09-21-55.03d4582f-ed06-4268-a70b-0dd9307fb4cf.wav)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *2 record_session::/usr/local/freeswitch/var/lib/freeswitch/recordings/domain.example.com/2016-02-13-09-21-55.03d4582f-ed06-4268-a70b-0dd9307fb4cf.wav
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 bind_meta_app(3 b s execute_extension::cf XML features)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *3 execute_extension::cf XML features
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 bind_meta_app(4 b s execute_extension::attented_xfer XML default)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *4 execute_extension::attented_xfer XML default
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 bind_meta_app(5 b s execute_extension::attented_xfer_to_vm XML default)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *5 execute_extension::attented_xfer_to_vm XML default
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 bind_meta_app(6 b s execute_extension::blind_xfer_to XML default)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *6 execute_extension::blind_xfer_to XML default
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 bind_meta_app(9 b s execute_extension::call_park XML default)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *9 execute_extension::call_park XML default
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 export(sip_contact_user=142)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [sip_contact_user]=[142]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(ringback=%(2000,4000,440,480))
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [ringback]=[%(2000,4000,440,480)]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(transfer_ringback=local_stream://moh)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [transfer_ringback]=[local_stream://moh]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(call_timeout=35)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [call_timeout]=[35]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(hangup_after_bridge=true)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [hangup_after_bridge]=[true]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 bridge(user/1276 at domain.example.com)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [domain_name]=[domain.example.com] to event
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [caller-id-in-from]=[true] to event
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [sip_cid_type]=[none] to event
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [domain_name]=[domain.example.com] to event
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [skip_cdr_causes]=[LOSE_RACE] to event
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [accountcode]=[17] to event
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [dialed_extension]=[1276] to event
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [available_to_call]=[true] to event
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [RFC2822_DATE]=[Sat, 13 Feb 2016 09:21:55 -0800] to event
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [domain_name]=[domain.example.com] to event
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [sip_contact_user]=[142] to event
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_ivr_originate.c:2128 Parsing global variables
2016-02-13 09:21:55.677785 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f41ac03e390 Connected.
2016-02-13 09:21:55.677785 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f41ac03e390 released.
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [domain_name]=[domain.example.com] to event
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [caller-id-in-from]=[true] to event
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [sip_cid_type]=[none] to event
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [domain_name]=[domain.example.com] to event
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [skip_cdr_causes]=[LOSE_RACE] to event
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [accountcode]=[17] to event
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [dialed_extension]=[1276] to event
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [available_to_call]=[true] to event
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [RFC2822_DATE]=[Sat, 13 Feb 2016 09:21:55 -0800] to event
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [domain_name]=[domain.example.com] to event
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [sip_contact_user]=[142] to event
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_ivr_originate.c:2128 Parsing global variables
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/1276 at 192.168.1.168:12113 [997ba225-1002-4fec-9b89-f7e1986ebfca]
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] mod_sofia.c:4776 (sofia/internal/1276 at 192.168.1.168:12113) State Change CS_NEW -> CS_INIT
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at 192.168.1.168:12113) Running State Change CS_INIT
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/1276 at 192.168.1.168:12113) State INIT
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] mod_sofia.c:88 sofia/internal/1276 at 192.168.1.168:12113 SOFIA INIT
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] sofia_glue.c:1228 sip:1276 at 68.5.94.63:12113 Setting proxy route to sofia/internal/1276 at 192.168.1.168:12113
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] sofia_glue.c:1257 sofia/internal/1276 at 192.168.1.168:12113 sending invite version: 1.6.6  64bit
997ba225-1002-4fec-9b89-f7e1986ebfca Local SDP:
997ba225-1002-4fec-9b89-f7e1986ebfca v=0
997ba225-1002-4fec-9b89-f7e1986ebfca o=FreeSWITCH 1455351947 1455351948 IN IP4 219.206.20.22
997ba225-1002-4fec-9b89-f7e1986ebfca s=FreeSWITCH
997ba225-1002-4fec-9b89-f7e1986ebfca c=IN IP4 219.206.20.22
997ba225-1002-4fec-9b89-f7e1986ebfca t=0 0
997ba225-1002-4fec-9b89-f7e1986ebfca m=audio 32168 RTP/AVP 0 8 101 13
997ba225-1002-4fec-9b89-f7e1986ebfca a=rtpmap:0 PCMU/8000
997ba225-1002-4fec-9b89-f7e1986ebfca a=rtpmap:8 PCMA/8000
997ba225-1002-4fec-9b89-f7e1986ebfca a=rtpmap:101 telephone-event/8000
997ba225-1002-4fec-9b89-f7e1986ebfca a=fmtp:101 0-16
997ba225-1002-4fec-9b89-f7e1986ebfca a=rtpmap:13 CN/8000
997ba225-1002-4fec-9b89-f7e1986ebfca a=ptime:20
997ba225-1002-4fec-9b89-f7e1986ebfca a=sendrecv
997ba225-1002-4fec-9b89-f7e1986ebfca 
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] switch_core_state_machine.c:40 sofia/internal/1276 at 192.168.1.168:12113 Standard INIT
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/1276 at 192.168.1.168:12113) State Change CS_INIT -> CS_ROUTING
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/1276 at 192.168.1.168:12113) State INIT going to sleep
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at 192.168.1.168:12113) Running State Change CS_ROUTING
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] sofia.c:6760 Channel sofia/internal/1276 at 192.168.1.168:12113 entering state [calling][0]
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/1276 at 192.168.1.168:12113) State ROUTING
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] mod_sofia.c:141 sofia/internal/1276 at 192.168.1.168:12113 SOFIA ROUTING
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/1276 at 192.168.1.168:12113) State Change CS_ROUTING -> CS_CONSUME_MEDIA
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/1276 at 192.168.1.168:12113) State ROUTING going to sleep
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at 192.168.1.168:12113) Running State Change CS_CONSUME_MEDIA
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] switch_core_state_machine.c:551 (sofia/internal/1276 at 192.168.1.168:12113) State CONSUME_MEDIA
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] switch_core_state_machine.c:551 (sofia/internal/1276 at 192.168.1.168:12113) State CONSUME_MEDIA going to sleep
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [ALERT] switch_core_state_machine.c:590 sofia/internal/1276 at 192.168.1.168:12113 session thread sleep state: CS_CONSUME_MEDIA!
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.907833 [ALERT] switch_core_state_machine.c:594 sofia/internal/1276 at 192.168.1.168:12113 session thread wake state: CS_CONSUME_MEDIA!
2016-02-13 09:21:55.907833 [ALERT] sofia.c:1214 sofia/internal/1276 at 192.168.1.168:12113 Same Callee ID "Outbound Call" <1276>
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.907833 [DEBUG] sofia.c:6760 Channel sofia/internal/1276 at 192.168.1.168:12113 entering state [proceeding][180]
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.907833 [NOTICE] sofia.c:6862 Ring-Ready sofia/internal/1276 at 192.168.1.168:12113!
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.907833 [DEBUG] switch_channel.c:3340 (sofia/internal/1276 at 192.168.1.168:12113) Callstate Change DOWN -> RINGING
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.907833 [ALERT] sofia.c:6862 sofia/internal/1276 at 192.168.1.168:12113 receive message [RING_EVENT]
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.907833 [ALERT] switch_core_state_machine.c:590 sofia/internal/1276 at 192.168.1.168:12113 session thread sleep state: CS_CONSUME_MEDIA!
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [ALERT] switch_ivr_originate.c:1216 sofia/internal/142 at domain.example.com:5660 receive message [PROGRESS]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [INFO] switch_ivr_originate.c:1216 Sending early media
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [ALERT] switch_core_media.c:413 Looking for zrtp-hash
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [ALERT] switch_core_media.c:368 Deciding whether to pass zrtp-hash between legs
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [ALERT] switch_core_media.c:376 Found peer channel; propagating zrtp-hash if set
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [ALERT] switch_core_media.c:303 Deciding whether to pass zrtp-hash between a-leg and b-leg
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [ALERT] switch_core_media.c:303 Deciding whether to pass zrtp-hash between a-leg and b-leg
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G729:18:8000:20:8000:1]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G729:18:8000:20:8000:1]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:4077 Set telephone-event payload to 101 at 8000
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:2906 Set Codec sofia/internal/142 at domain.example.com:5660 PCMU/8000 20 ms 160 samples 64000 bits 1 channels
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_codec.c:111 sofia/internal/142 at domain.example.com:5660 Original read codec set to PCMU:0
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:4429 Set telephone-event payload to 101 at 8000
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:4485 sofia/internal/142 at domain.example.com:5660 Set 2833 dtmf send payload to 101 recv payload to 101
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:6033 AUDIO RTP [sofia/internal/142 at domain.example.com:5660] 219.206.20.22 port 30674 -> 192.168.33.116 port 50028 codec: 0 ms: 20
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_rtp.c:3802 Starting timer [soft] 160 bytes per 20ms
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:1939 Setting Jitterbuffer to 60ms (3 frames) (50 max frames)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:6332 sofia/internal/142 at domain.example.com:5660 Set 2833 dtmf send payload to 101
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:6339 sofia/internal/142 at domain.example.com:5660 Set 2833 dtmf receive payload to 101
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:6362 sofia/internal/142 at domain.example.com:5660 Set rtp dtmf delay to 60
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/142 at domain.example.com:5660!
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_channel.c:3468 (sofia/internal/142 at domain.example.com:5660) Callstate Change RINGING -> EARLY
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [ALERT] sofia_media.c:92 sofia/internal/142 at domain.example.com:5660 receive message [PROGRESS_EVENT]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] mod_sofia.c:2330 Ring SDP:
03d4582f-ed06-4268-a70b-0dd9307fb4cf v=0
03d4582f-ed06-4268-a70b-0dd9307fb4cf o=FreeSWITCH 1455353441 1455353442 IN IP4 219.206.20.22
03d4582f-ed06-4268-a70b-0dd9307fb4cf s=FreeSWITCH
03d4582f-ed06-4268-a70b-0dd9307fb4cf c=IN IP4 219.206.20.22
03d4582f-ed06-4268-a70b-0dd9307fb4cf t=0 0
03d4582f-ed06-4268-a70b-0dd9307fb4cf m=audio 30674 RTP/AVP 0 101
03d4582f-ed06-4268-a70b-0dd9307fb4cf a=rtpmap:0 PCMU/8000
03d4582f-ed06-4268-a70b-0dd9307fb4cf a=rtpmap:101 telephone-event/8000
03d4582f-ed06-4268-a70b-0dd9307fb4cf a=fmtp:101 0-16
03d4582f-ed06-4268-a70b-0dd9307fb4cf a=ptime:20
03d4582f-ed06-4268-a70b-0dd9307fb4cf a=sendrecv
03d4582f-ed06-4268-a70b-0dd9307fb4cf 
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_ivr_originate.c:1274 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_codec.c:221 sofia/internal/142 at domain.example.com:5660 Push codec L16:100
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_ivr_originate.c:1343 Play Ringback Tone [%(2000,4000,440,480)]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.947831 [DEBUG] sofia.c:6760 Channel sofia/internal/142 at domain.example.com:5660 entering state [early][183]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.947831 [ALERT] switch_channel.c:3507 sofia/internal/142 at domain.example.com:5660 receive message [AUDIO_SYNC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.947831 [ALERT] switch_core_io.c:1135 sofia/internal/142 at domain.example.com:5660 receive message [TRANSCODING_NECESSARY]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:56.817831 [INFO] switch_rtp.c:6616 Auto Changing audio port from 192.168.33.116:50028 to 73.170.102.43:50028
2016-02-13 09:22:00.847852 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f41ac03e390 Connected.
2016-02-13 09:22:00.847852 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f41ac03e390 released.
2016-02-13 09:22:03.957850 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f41ac03e390 Connected.
2016-02-13 09:22:03.957850 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f41ac03e390 released.
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_state_machine.c:594 sofia/internal/1276 at 192.168.1.168:12113 session thread wake state: CS_CONSUME_MEDIA!
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_media.c:413 Looking for zrtp-hash
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_media.c:368 Deciding whether to pass zrtp-hash between legs
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_media.c:376 Found peer channel; propagating zrtp-hash if set
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_media.c:303 Deciding whether to pass zrtp-hash between a-leg and b-leg
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_media.c:303 Deciding whether to pass zrtp-hash between a-leg and b-leg
2016-02-13 09:22:04.037826 [ALERT] sofia.c:1214 sofia/internal/1276 at 192.168.1.168:12113 Same Callee ID "Outbound Call" <1276>
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] sofia.c:6760 Channel sofia/internal/1276 at 192.168.1.168:12113 entering state [completing][200]
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] sofia.c:6770 Remote SDP:
997ba225-1002-4fec-9b89-f7e1986ebfca v=0
997ba225-1002-4fec-9b89-f7e1986ebfca o=1276 82204 1 IN IP4 192.168.1.168
997ba225-1002-4fec-9b89-f7e1986ebfca s=-
997ba225-1002-4fec-9b89-f7e1986ebfca c=IN IP4 192.168.1.168
997ba225-1002-4fec-9b89-f7e1986ebfca t=0 0
997ba225-1002-4fec-9b89-f7e1986ebfca m=audio 8000 RTP/AVP 0 96
997ba225-1002-4fec-9b89-f7e1986ebfca a=rtpmap:0 PCMU/8000
997ba225-1002-4fec-9b89-f7e1986ebfca a=rtpmap:96 telephone-event/8000
997ba225-1002-4fec-9b89-f7e1986ebfca a=ptime:20
997ba225-1002-4fec-9b89-f7e1986ebfca 
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_state_machine.c:590 sofia/internal/1276 at 192.168.1.168:12113 session thread sleep state: CS_CONSUME_MEDIA!
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_state_machine.c:594 sofia/internal/1276 at 192.168.1.168:12113 session thread wake state: CS_CONSUME_MEDIA!
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] sofia.c:6760 Channel sofia/internal/1276 at 192.168.1.168:12113 entering state [ready][200]
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_media.c:413 Looking for zrtp-hash
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_media.c:368 Deciding whether to pass zrtp-hash between legs
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_media.c:376 Found peer channel; propagating zrtp-hash if set
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_media.c:303 Deciding whether to pass zrtp-hash between a-leg and b-leg
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_media.c:303 Deciding whether to pass zrtp-hash between a-leg and b-leg
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_core_media.c:4077 Set telephone-event payload to 96 at 8000
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_core_media.c:2906 Set Codec sofia/internal/1276 at 192.168.1.168:12113 PCMU/8000 20 ms 160 samples 64000 bits 1 channels
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_core_codec.c:111 sofia/internal/1276 at 192.168.1.168:12113 Original read codec set to PCMU:0
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_core_media.c:4429 Set telephone-event payload to 96 at 8000
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_core_media.c:4485 sofia/internal/1276 at 192.168.1.168:12113 Set 2833 dtmf send payload to 96 recv payload to 101
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_core_media.c:6033 AUDIO RTP [sofia/internal/1276 at 192.168.1.168:12113] 219.206.20.22 port 32168 -> 192.168.1.168 port 8000 codec: 0 ms: 20
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_rtp.c:3802 Starting timer [soft] 160 bytes per 20ms
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_core_media.c:1939 Setting Jitterbuffer to 60ms (3 frames) (50 max frames)
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_core_media.c:6332 sofia/internal/1276 at 192.168.1.168:12113 Set 2833 dtmf send payload to 96
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_core_media.c:6339 sofia/internal/1276 at 192.168.1.168:12113 Set 2833 dtmf receive payload to 101
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_core_media.c:6362 sofia/internal/1276 at 192.168.1.168:12113 Set rtp dtmf delay to 60
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [NOTICE] sofia.c:7724 Channel [sofia/internal/1276 at 192.168.1.168:12113] has been answered
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_channel.c:3767 (sofia/internal/1276 at 192.168.1.168:12113) Callstate Change RINGING -> ACTIVE
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] sofia.c:7724 sofia/internal/1276 at 192.168.1.168:12113 receive message [ANSWER_EVENT]
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_state_machine.c:590 sofia/internal/1276 at 192.168.1.168:12113 session thread sleep state: CS_CONSUME_MEDIA!
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [DEBUG] switch_core_codec.c:246 sofia/internal/142 at domain.example.com:5660 Restore previous codec PCMU:0.
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [ALERT] switch_ivr_originate.c:3550 sofia/internal/142 at domain.example.com:5660 receive message [ANSWER]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [DEBUG] mod_sofia.c:799 Local SDP sofia/internal/142 at domain.example.com:5660:
03d4582f-ed06-4268-a70b-0dd9307fb4cf v=0
03d4582f-ed06-4268-a70b-0dd9307fb4cf o=FreeSWITCH 1455353441 1455353443 IN IP4 219.206.20.22
03d4582f-ed06-4268-a70b-0dd9307fb4cf s=FreeSWITCH
03d4582f-ed06-4268-a70b-0dd9307fb4cf c=IN IP4 219.206.20.22
03d4582f-ed06-4268-a70b-0dd9307fb4cf t=0 0
03d4582f-ed06-4268-a70b-0dd9307fb4cf m=audio 30674 RTP/AVP 0 101
03d4582f-ed06-4268-a70b-0dd9307fb4cf a=rtpmap:0 PCMU/8000
03d4582f-ed06-4268-a70b-0dd9307fb4cf a=rtpmap:101 telephone-event/8000
03d4582f-ed06-4268-a70b-0dd9307fb4cf a=fmtp:101 0-16
03d4582f-ed06-4268-a70b-0dd9307fb4cf a=ptime:20
03d4582f-ed06-4268-a70b-0dd9307fb4cf a=sendrecv
03d4582f-ed06-4268-a70b-0dd9307fb4cf 
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [NOTICE] switch_ivr_originate.c:3550 Channel [sofia/internal/142 at domain.example.com:5660] has been answered
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [DEBUG] switch_channel.c:3767 (sofia/internal/142 at domain.example.com:5660) Callstate Change EARLY -> ACTIVE
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [ALERT] switch_ivr_originate.c:3550 sofia/internal/142 at domain.example.com:5660 receive message [ANSWER_EVENT]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [DEBUG] sofia.c:6760 Channel sofia/internal/142 at domain.example.com:5660 entering state [completed][200]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [DEBUG] switch_ivr_originate.c:3608 Originate Resulted in Success: [sofia/internal/1276 at 192.168.1.168:12113]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [ALERT] switch_ivr_originate.c:3944 sofia/internal/142 at domain.example.com:5660 receive message [AUDIO_SYNC]
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [ALERT] switch_core_state_machine.c:594 sofia/internal/1276 at 192.168.1.168:12113 session thread wake state: CS_CONSUME_MEDIA!
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [ALERT] switch_ivr_originate.c:3941 sofia/internal/1276 at 192.168.1.168:12113 receive message [AUDIO_SYNC]
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [ALERT] switch_core_state_machine.c:590 sofia/internal/1276 at 192.168.1.168:12113 session thread sleep state: CS_CONSUME_MEDIA!
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [DEBUG] switch_ivr_originate.c:3608 Originate Resulted in Success: [sofia/internal/1276 at 192.168.1.168:12113]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [ALERT] switch_ivr_originate.c:3944 sofia/internal/142 at domain.example.com:5660 receive message [AUDIO_SYNC]
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [ALERT] switch_core_state_machine.c:594 sofia/internal/1276 at 192.168.1.168:12113 session thread wake state: CS_CONSUME_MEDIA!
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [ALERT] switch_ivr_originate.c:3941 sofia/internal/1276 at 192.168.1.168:12113 receive message [AUDIO_SYNC]
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [ALERT] switch_core_state_machine.c:590 sofia/internal/1276 at 192.168.1.168:12113 session thread sleep state: CS_CONSUME_MEDIA!
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [ALERT] switch_core_state_machine.c:594 sofia/internal/1276 at 192.168.1.168:12113 session thread wake state: CS_CONSUME_MEDIA!
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [ALERT] switch_ivr_bridge.c:1452 sofia/internal/1276 at 192.168.1.168:12113 receive message [AUDIO_SYNC]
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [ALERT] switch_core_state_machine.c:590 sofia/internal/1276 at 192.168.1.168:12113 session thread sleep state: CS_CONSUME_MEDIA!
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [ALERT] switch_ivr.c:217 sofia/internal/142 at domain.example.com:5660 receive message [AUDIO_SYNC]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [ALERT] switch_ivr_bridge.c:1451 sofia/internal/142 at domain.example.com:5660 receive message [AUDIO_SYNC]
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [ALERT] switch_ivr_bridge.c:1551 sofia/internal/1276 at 192.168.1.168:12113 receive message [BRIDGE]
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [DEBUG] switch_core_media.c:9156 sofia/internal/1276 at 192.168.1.168:12113 PAUSE Jitterbuffer
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [ALERT] switch_ivr_bridge.c:1559 sofia/internal/142 at domain.example.com:5660 receive message [BRIDGE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [DEBUG] switch_core_media.c:9156 sofia/internal/142 at domain.example.com:5660 PAUSE Jitterbuffer
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [DEBUG] switch_ivr_bridge.c:1591 (sofia/internal/1276 at 192.168.1.168:12113) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [ALERT] switch_core_state_machine.c:594 sofia/internal/1276 at 192.168.1.168:12113 session thread wake state: CS_CONSUME_MEDIA!
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at 192.168.1.168:12113) Running State Change CS_EXCHANGE_MEDIA
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [DEBUG] switch_core_state_machine.c:542 (sofia/internal/1276 at 192.168.1.168:12113) State EXCHANGE_MEDIA
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [DEBUG] mod_sofia.c:613 SOFIA EXCHANGE_MEDIA
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.067829 [ALERT] switch_core_io.c:420 sofia/internal/142 at domain.example.com:5660 receive message [TRANSCODING_NECESSARY]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.087830 [DEBUG] switch_rtp.c:6654 Correct audio ip/port confirmed.
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.257832 [INFO] switch_rtp.c:6616 Auto Changing audio port from 192.168.1.168:8000 to 68.5.94.63:8000
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.587826 [DEBUG] sofia.c:6760 Channel sofia/internal/142 at domain.example.com:5660 entering state [ready][200]
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.597832 [ALERT] switch_ivr_bridge.c:283 sofia/internal/1276 at 192.168.1.168:12113 receive message [DISPLAY]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.607833 [ALERT] switch_ivr_bridge.c:283 sofia/internal/142 at domain.example.com:5660 receive message [DISPLAY]
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:06.197829 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at 192.168.1.168:12113 Got: audio seq 53854 but expected: 53852 lost: 2
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:06.617854 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at 192.168.1.168:12113 Got: audio seq 53866 but expected: 53865 lost: 1

997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:08.737851 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at 192.168.1.168:12113 Got: audio seq 53961 but expected: 53960 lost: 1
2016-02-13 09:22:08.827825 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f41ac03e390 Connected.
2016-02-13 09:22:08.827825 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f41ac03e390 released.
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:10.097827 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at 192.168.1.168:12113 Got: audio seq 53977 but expected: 53974 lost: 3
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:10.637831 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at 192.168.1.168:12113 Got: audio seq 53990 but expected: 53989 lost: 1
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:11.717833 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at 192.168.1.168:12113 Got: audio seq 54036 but expected: 54035 lost: 1
2016-02-13 09:22:11.967873 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f41ac03e390 Connected.
2016-02-13 09:22:11.967873 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f41ac03e390 released.
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.107862 [NOTICE] sofia.c:952 Hangup sofia/internal/142 at domain.example.com:5660 [CS_EXECUTE] [NORMAL_CLEARING]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.107862 [DEBUG] switch_ivr_bridge.c:778 BRIDGE THREAD DONE [sofia/internal/142 at domain.example.com:5660]
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_ivr_bridge.c:699 sofia/internal/142 at domain.example.com:5660 ending bridge by request from write function
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_ivr_bridge.c:778 BRIDGE THREAD DONE [sofia/internal/1276 at 192.168.1.168:12113]
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [NOTICE] switch_ivr_bridge.c:881 Hangup sofia/internal/1276 at 192.168.1.168:12113 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:542 (sofia/internal/1276 at 192.168.1.168:12113) State EXCHANGE_MEDIA going to sleep
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at 192.168.1.168:12113) Running State Change CS_HANGUP
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:739 (sofia/internal/1276 at 192.168.1.168:12113) Callstate Change ACTIVE -> HANGUP
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/1276 at 192.168.1.168:12113) State HANGUP
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] mod_sofia.c:425 sofia/internal/1276 at 192.168.1.168:12113 Overriding SIP cause 480 with 200 from the other leg
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] mod_sofia.c:431 Channel sofia/internal/1276 at 192.168.1.168:12113 hanging up, cause: NORMAL_CLEARING
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] mod_sofia.c:484 Sending BYE to sofia/internal/1276 at 192.168.1.168:12113
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1276 at 192.168.1.168:12113 Standard HANGUP, cause: NORMAL_CLEARING
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/1276 at 192.168.1.168:12113) State HANGUP going to sleep
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [ALERT] switch_ivr_bridge.c:1689 sofia/internal/1276 at 192.168.1.168:12113 receive message [UNBRIDGE]
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_ivr_bridge.c:1689 sofia/internal/1276 at 192.168.1.168:12113 skip receive message [UNBRIDGE] (channel is hungup already)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [ALERT] switch_ivr_bridge.c:1692 sofia/internal/142 at domain.example.com:5660 receive message [UNBRIDGE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [DEBUG] switch_ivr_bridge.c:1692 sofia/internal/142 at domain.example.com:5660 skip receive message [UNBRIDGE] (channel is hungup already)
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:508 (sofia/internal/1276 at 192.168.1.168:12113) State Change CS_HANGUP -> CS_REPORTING
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at 192.168.1.168:12113) Running State Change CS_REPORTING
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [DEBUG] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:539 (sofia/internal/142 at domain.example.com:5660) State EXECUTE going to sleep
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at domain.example.com:5660) Running State Change CS_HANGUP
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/1276 at 192.168.1.168:12113) State REPORTING
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:104 sofia/internal/1276 at 192.168.1.168:12113 Standard REPORTING, cause: NORMAL_CLEARING
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/1276 at 192.168.1.168:12113) State REPORTING going to sleep
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:739 (sofia/internal/142 at domain.example.com:5660) Callstate Change ACTIVE -> HANGUP
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/1276 at 192.168.1.168:12113) State Change CS_REPORTING -> CS_DESTROY
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_session.c:1646 Session 1855 (sofia/internal/1276 at 192.168.1.168:12113) Locked, Waiting on external entities
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [NOTICE] switch_core_session.c:1664 Session 1855 (sofia/internal/1276 at 192.168.1.168:12113) Ended
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [NOTICE] switch_core_session.c:1668 Close Channel sofia/internal/1276 at 192.168.1.168:12113 [CS_DESTROY]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/142 at domain.example.com:5660) State HANGUP
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [DEBUG] mod_sofia.c:431 Channel sofia/internal/142 at domain.example.com:5660 hanging up, cause: NORMAL_CLEARING
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:630 (sofia/internal/1276 at 192.168.1.168:12113) Running State Change CS_DESTROY
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/1276 at 192.168.1.168:12113) State DESTROY
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] mod_sofia.c:341 sofia/internal/1276 at 192.168.1.168:12113 SOFIA DESTROY
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:111 sofia/internal/1276 at 192.168.1.168:12113 Standard DESTROY
997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/1276 at 192.168.1.168:12113) State DESTROY going to sleep
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:60 sofia/internal/142 at domain.example.com:5660 Standard HANGUP, cause: NORMAL_CLEARING
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/142 at domain.example.com:5660) State HANGUP going to sleep
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:508 (sofia/internal/142 at domain.example.com:5660) State Change CS_HANGUP -> CS_REPORTING
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at domain.example.com:5660) Running State Change CS_REPORTING
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/142 at domain.example.com:5660) State REPORTING
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [DEBUG] mod_odbc_cdr.c:309 sql INSERT INTO cebod_cdr (used_gateway_id, phonenumber, call_uuid, leg, job_uuid, callrequest_id, accountcode, created_date, callerid, billsec, callername, starting_date, hangup_cause, duration, event_name) VALUES (73.170.102.43, 1276, 03d4582f-ed06-4268-a70b-0dd9307fb4cf, inbound, 03d4582f-ed06-4268-a70b-0dd9307fb4cf, 03d4582f-ed06-4268-a70b-0dd9307fb4cf, 17, 2016-02-13 09:22:12, 142, 8, 142, 2016-02-13 09:21:55, NORMAL_CLEARING, 17, REQUEST_PARAMS)
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.127828 [DEBUG] switch_core_state_machine.c:104 sofia/internal/142 at domain.example.com:5660 Standard REPORTING, cause: NORMAL_CLEARING
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.127828 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/142 at domain.example.com:5660) State REPORTING going to sleep
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.127828 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/142 at domain.example.com:5660) State Change CS_REPORTING -> CS_DESTROY
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.127828 [DEBUG] switch_core_session.c:1646 Session 1854 (sofia/internal/142 at domain.example.com:5660) Locked, Waiting on external entities
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.127828 [NOTICE] switch_core_session.c:1664 Session 1854 (sofia/internal/142 at domain.example.com:5660) Ended
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.127828 [NOTICE] switch_core_session.c:1668 Close Channel sofia/internal/142 at domain.example.com:5660 [CS_DESTROY]
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.127828 [DEBUG] switch_core_state_machine.c:630 (sofia/internal/142 at domain.example.com:5660) Running State Change CS_DESTROY
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.127828 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/142 at domain.example.com:5660) State DESTROY
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.127828 [DEBUG] mod_sofia.c:341 sofia/internal/142 at domain.example.com:5660 SOFIA DESTROY
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.127828 [DEBUG] switch_core_state_machine.c:111 sofia/internal/142 at domain.example.com:5660 Standard DESTROY
03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.127828 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/142 at domain.example.com:5660) State DESTROY going to sleep

From rajil.s at gmail.com  Sun Feb 14 05:53:54 2016
From: rajil.s at gmail.com (Rajil Saraswat)
Date: Sat, 13 Feb 2016 20:53:54 -0600
Subject: [Freeswitch-users] Freeswitch doesnt transcode
Message-ID: <CAFoaQoAoVLu7ni7x02Y-2hS3nvjf_vdphJkLqt4+mxbhx7=g+w@mail.gmail.com>

Hello,

I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
When I call to this remote sip phone i get a 406 error that opus is
not supported as shown by the sip trace below. However, if I force the
codec to absolute like this
{absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5
the call works fine.

Is there anyway I can make FreeSWITCH to automatically transcode
without forcing the codec string in the dial plan?

The codec preferences is set as  global_codec_prefs=OPUS,PCMU,PCMA and
outbound_codec_prefs=PCMU,PCMA,GSM

---------------------------siptrace--------------------------------

recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
   ------------------------------------------------------------------------
   SIP/2.0 406 Not Acceptable
   Via: SIP/2.0/UDP
192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
   From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
   To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
   CSeq: 87372504 INVITE
   Content-Length:  0

   ------------------------------------------------------------------------
send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:
   ------------------------------------------------------------------------
   ACK sip:303 at 192.168.1.5 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
   Max-Forwards: 68
   From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
   To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
   CSeq: 87372504 ACK
   Content-Length: 0

   ------------------------------------------------------------------------
2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
sofia/internal/303 at 192.168.1.5 entering state [terminated][406]
2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA]
[SERVICE_NOT_IMPLEMENTED]

Thanks
Rajil


From dave at dchorton.com  Sun Feb 14 06:49:04 2016
From: dave at dchorton.com (Dave Horton)
Date: Sat, 13 Feb 2016 22:49:04 -0500
Subject: [Freeswitch-users] Oneway audio issues in freeswitch
In-Reply-To: <CAPfjz3jwe74Vvz1515TQxOJdAYsNuRXA_CTtzTPfG9EjuXXhEA@mail.gmail.com>
References: <CANw+5nWmd_j=eUz7OGoH=5Wb3g43VOS_XF_dZVkuFRMRUtfZ2Q@mail.gmail.com>
	<CABMX_w_NWvQ5ZUdxtYHnua4FD9i4hYDt3YeP-X+43XYi4cb=8A@mail.gmail.com>
	<CAFLo2QzdR1YBLv6-y5hs=juyE5N2uZw+N5dCiyq4S3Hbgkb2eg@mail.gmail.com>
	<CANw+5nUWV3dmJGUtDgLBSaUuGufcXskoYNyj6FJcmd+yRs184g@mail.gmail.com>
	<CAPcNkdfppepyyC+OLTr4OFyhFWhP9MjWR0WUsraU-ABrcJLY1Q@mail.gmail.com>
	<CAPfjz3jwe74Vvz1515TQxOJdAYsNuRXA_CTtzTPfG9EjuXXhEA@mail.gmail.com>
Message-ID: <67F80D46-E511-459C-A72B-E966671D4AB3@dchorton.com>

Redo with sofia global siptrace on so we can see the SIP messaging
On Feb 13, 2016, at 12:45 PM, Jai Rangi <jprangi at didforsale.com> wrote:

Hello Ken,
Thank for look in this. Attached are debug logs. SIP Traces were not molested, except the public IPs were changed. As of writing of this email, the issue is isolated to 1.6.x. 
Not sure if anyone else has tested this on latest version. But easy to reproduce. Just download grandstream Wave, available to IOS and Andriod and try to call any extension directly. Curious to see if any one can come with different result. 

Jai Rangi
Cebod Technologies LLC dba DIDforSale/Cebod Telecom
O 949-471-0102 <tel:O%20949-471-0101> | C 949-419-7634 <tel:C%20949-742-2666> | F 949-269-0449 / 949-232-1410 <> | jprangi at didforsale.com <mailto:jprangi at didforsale.com>  www.cebod.com <http://www.cebod.com/> |  www.didforsale.com <http://www.didforsale.com/> |3200 Bristol St Suite 615, Costa Mesa, CA 92626  | 






On Thu, Feb 11, 2016 at 8:36 AM, Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org>> wrote:
without logs of a call doing this at debug level with a complete unmolested sip trace in line its a little hard to speculate whats going on here

On Thu, Feb 11, 2016 at 10:28 AM, mohammed shafeeque <shafeeq.v at gmail.com <mailto:shafeeq.v at gmail.com>> wrote:
Surprised that no one else experienced this problem. Can anyone give any hint. Really Dont want to move back to 1.4.x

On Thu, Feb 11, 2016 at 7:44 AM, Luis Daniel Lucio Quiroz <luis.daniel.lucio at gmail.com <mailto:luis.daniel.lucio at gmail.com>> wrote:
As a rule of dumb, try turning on rport

Le 10 f?vr. 2016 8:55 PM, "?talo Rossi" <italo at freeswitch.org <mailto:italo at freeswitch.org>> a ?crit :
You need to look at the sip signaling to see what's going on

On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque <shafeeq.v at gmail.com <mailto:shafeeq.v at gmail.com>> wrote:
Hello All

We are getting one way audio issues with some softphones and grandstream phones behind nat registerd to our freeswitch server. 

Here is scenario: 
Grandstream call any extensions (one way audio) 
Any extension call Grandstream ( Audio works just fine)

We have tried multiple softphones and the result is same. 

Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started with an upgrade to freeswitch. 

Any help or hint will be much appreciated. 

Thank you, 


_________________________________________________________________________
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/>
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http://www.cluecon.com <http://www.cluecon.com/>

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-- 
?talo Rossi
italo at freeswitch.org <mailto:italo at 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/>

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_________________________________________________________________________
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Official FreeSWITCH Sites
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_________________________________________________________________________
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
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_________________________________________________________________________
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/>
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http://www.cluecon.com <http://www.cluecon.com/>

FreeSWITCH-users mailing list
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<freeswitch.log.goodcall.txt><freeswitch.log.badcall.txt>_________________________________________________________________________
Professional FreeSWITCH Consulting Services: 
consulting at freeswitch.org
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From vbvbrj at gmail.com  Sun Feb 14 10:35:07 2016
From: vbvbrj at gmail.com (Mimiko)
Date: Sun, 14 Feb 2016 09:35:07 +0200
Subject: [Freeswitch-users] Fwd: hook to event or startup script which
 will hook to events?
In-Reply-To: <CAO=waFCbwKiAOnVyxSW-deT3GS7K2wGH4ZsV8H139aVUB2XpUQ@mail.gmail.com>
References: <56B75BDE.6020604@gmail.com>
	<CAO=waFCbwKiAOnVyxSW-deT3GS7K2wGH4ZsV8H139aVUB2XpUQ@mail.gmail.com>
Message-ID: <56C02E2B.2090304@gmail.com>

On 14.02.2016 02:38, Abdul Hakeem wrote:
> Hello Mimiko,
> Did you find any resolution to the issue ?
> Cheers,
> AH


No, no answers to my questions.

I use start-up script in lua.conf and bind to event in order to avoid 
segfault.

-- 
Mimiko desu.


From idokan at gmail.com  Sun Feb 14 17:16:44 2016
From: idokan at gmail.com (ik)
Date: Sun, 14 Feb 2016 16:16:44 +0200
Subject: [Freeswitch-users] Fwd: hook to event or startup script which
 will hook to events?
In-Reply-To: <56C02E2B.2090304@gmail.com>
References: <56B75BDE.6020604@gmail.com>
	<CAO=waFCbwKiAOnVyxSW-deT3GS7K2wGH4ZsV8H139aVUB2XpUQ@mail.gmail.com>
	<56C02E2B.2090304@gmail.com>
Message-ID: <CAOccsKB+rB8OKsRgX8YZSK0PV-sFYMs324zGGrr6V5n901=SVg@mail.gmail.com>

I'ved created a Perl daemon for capturing specific events.

You can filter to do so.

Ido
On Feb 14, 2016 9:37 AM, "Mimiko" <vbvbrj at gmail.com> wrote:

> On 14.02.2016 02:38, Abdul Hakeem wrote:
> > Hello Mimiko,
> > Did you find any resolution to the issue ?
> > Cheers,
> > AH
>
>
> No, no answers to my questions.
>
> I use start-up script in lua.conf and bind to event in order to avoid
> segfault.
>
> --
> Mimiko desu.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From krice at freeswitch.org  Sun Feb 14 18:44:21 2016
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 14 Feb 2016 09:44:21 -0600
Subject: [Freeswitch-users] Fwd: hook to event or startup script which
	will hook to events?
In-Reply-To: <56C02E2B.2090304@gmail.com>
References: <56B75BDE.6020604@gmail.com>	<CAO=waFCbwKiAOnVyxSW-deT3GS7K2wGH4ZsV8H139aVUB2XpUQ@mail.gmail.com>
	<56C02E2B.2090304@gmail.com>
Message-ID: <7ee401d1673e$93284c20$b978e460$@freeswitch.org>

Is there a ticket open on this segfault you are avoiding and how to
reproduce it? 

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mimiko
Sent: Sunday, February 14, 2016 1:35 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Fwd: hook to event or startup script which
will hook to events?

On 14.02.2016 02:38, Abdul Hakeem wrote:
> Hello Mimiko,
> Did you find any resolution to the issue ?
> Cheers,
> AH


No, no answers to my questions.

I use start-up script in lua.conf and bind to event in order to avoid
segfault.

--
Mimiko desu.

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

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

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



From lists at kavun.ch  Sun Feb 14 19:18:16 2016
From: lists at kavun.ch (Emrah)
Date: Sun, 14 Feb 2016 17:18:16 +0100
Subject: [Freeswitch-users] High availability on different networks
In-Reply-To: <CAMbW9oJxHhWjcVvV7TdtzHMaxWGsu4L8kG-8YYmaNKXJPObgNQ@mail.gmail.com>
References: <78429801-FCA4-4BA9-8EA6-64A1BD7ECA28@kavun.ch>
	<CACOYK=gUYrGPouHq4voAygR7ctBOc6sdwBrAF8DWEeW6XkXS9A@mail.gmail.com>
	<CACOYK=jtfv356yOWXSJuqcVNL0L+_rMWAiQKEWqAC9i8bOwqUg@mail.gmail.com>
	<CAMbW9oJxHhWjcVvV7TdtzHMaxWGsu4L8kG-8YYmaNKXJPObgNQ@mail.gmail.com>
Message-ID: <1382DEBE-2DB3-48AC-AE54-5AD9EE370709@kavun.ch>

I love all your ideas, and thanks for sharing.
The best option remains to test things out in real scenarios and see what happens.
I?m already distributing my media traffic. Now I?d like to make the SIP part redundant. 
Just got some server resources in different locations. I?ll report back with my findings!

E

> On Feb 12, 2016, at 10:09 AM, Andrew Cassidy <andrew at cassidywebservices.co.uk> wrote:
> 
> It's not instant, but I've used OVH failover IP's to do that sort of thing. Heartbeat over ipsec tunnel, use pacemaker with a custom script that does the OVH API call to move the IP address.
> 
> Sadly it's not that quick, takes about 2 minutes.
> 
> On 12 February 2016 at 07:40, Stanislav Sinyagin <ssinyagin at gmail.com <mailto:ssinyagin at gmail.com>> wrote:
> there is an issue with anycast routing though: when you bring up a new
> server, some running sessions will be dropped, because their IP
> packets would be routed to a different host. So, it needs a careful
> design. Maybe place only the SIP proxy on an anycast address, and run
> the calls from unique local addresses.
> 
> 
> Multiple DNS SRV records with different priorities are also possible,
> but you can't direct the users to the nearest location within the same
> domain. Also a bunch of SIP clients needs to be tested and you need to
> answer the questions, like:
> 
> -- what is the timeout if the primary server is unavailable?
> -- if the primary host goes down during the call, how soon can the
> client re-dial?
> -- what happens if the primary server comes up again?
> 
> 
> 
> 
> 
> 
> On Thu, Feb 11, 2016 at 8:00 PM, Stanislav Sinyagin <ssinyagin at gmail.com <mailto:ssinyagin at gmail.com>> wrote:
> > hi Emrah and all,
> >
> > it's the first time I actually searched for it, but there are hosting
> > offers with anycast IP routing. It means, you have multiple servers in
> > various locations, and they share the same service IP address. The
> > clients connect to the nearest server, which is determined by standard
> > BGP routing. You are still limited to a single global hosting
> > provider, but you benefit from its redundant network and geographical
> > distribution.
> >
> > In case of anycast addressing, incoming connections will be served
> > easily. But the outgoing connections are rather tricky: you will need
> > to bring the outbound call to the physical server where the user has
> > registered, and initiate the connection from its anycast address. So,
> > you can share and replicate the registration database, but you need to
> > send the outbound call to the server which accepted the registration.
> > I guess you should be able to retrieve this information from the
> > registration database. This needs to be looked in details.
> >
> > Google for anycast server hosting, and there are at least 3 providers
> > offering virtual hosts, and OVH is offering physical hosts as well. I
> > guess there are more providers with similar offerings.
> >
> >
> > Without anycast, you would need to use redundant registrars sharing
> > the same service IP address -- for example, Digitalocean offers such
> > service within any single datacenter.
> >
> > Having multiple registrars with different IP addresses is also
> > possible, but then you depend on the way how each particular SIP
> > client handles multiple IP addresses after resolving the domain name.
> > Some of them may get stuck to a single address, even if it's not
> > responding.
> >
> >
> > cheers,
> > stanislav
> >
> > On Thu, Feb 11, 2016 at 5:53 PM, Emrah <lists at kavun.ch <mailto:lists at kavun.ch>> wrote:
> >> Hi list,
> >> I?m writing to gather your thoughts and suggestions on how to have a high availability FS setup on different networks.
> >>
> >> I am trying to achieve the following:
> >>  - Load balance FreeSWITCH instances on 2 or more servers, possibly in different countries.
> >> - Shared user directory and dialplan, but I?m not sure if shared registrations would make sense.
> >> - If a server goes down, the phone should register on the alternative servers. Obviously we can?t keep calls up.
> >>
> >> I?m obviously not the first one out there doing this. I?m trying to learn from those who?ve come up with reliable solutions.
> >>
> >> I?ve tried sharing a registration table among multiple FS instances. But it was a beginners mistake. Even with the right path to reach the client, only the invites sent from the server used by the phone would be processed.
> >> If my phone registers on server A, then server A shares the info with server B, server B knows how to contact the phone but it won?t be able to. Supposedly because of NAT issues.
> >>
> >> I am aiming for fully independent FS instances that can back each other up and be used independently. I am guessing this would require some sort of SBC or external registrar server with a Kamailio or Repro.
> >>
> >> Anyway just trying to spark the conversation around this subject and hopefully we can come up with a formula that can help many with their FS deployments. My provider?s network just went all down in IPv4 and HA behind the same provider proved to be useless.
> >>
> >> Best,
> >> E
> >> _________________________________________________________________________
> >> 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/>
> 
> 
> -- 
> Andrew Cassidy BSc (Hons) MBCS SSCA
> Managing Director
> 
> 
> T <mailto:info at cassidywebservices.co.uk> 03300 100 960  F <mailto:info at cassidywebservices.co.uk> 03300 100 961
> E <mailto:info at cassidywebservices.co.uk> andrew at cassidywebservices.co.uk <mailto:andrew at cassidywebservices.co.uk>
> W <mailto:info at cassidywebservices.co.uk> www.cassidywebservices.co.uk <http://www.cassidywebservices.co.uk/>_________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing 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 abaci64 at gmail.com  Sun Feb 14 19:41:47 2016
From: abaci64 at gmail.com (Abaci B)
Date: Sun, 14 Feb 2016 11:41:47 -0500
Subject: [Freeswitch-users] AMD
Message-ID: <CAFmZFiNO_yzv8vwO9HFM41v-B7w7mHwzBBcJ74wGmZxEm+Nq7g@mail.gmail.com>

Hi all,
I know that in general Intel Xeon processors are recommended over AMD for
FreeSWITCH, my question is if AMD is really bad or just not as good as
Intel, I can get now a server with 4x AMD 6174 (48 Core total) for about
the same price I would pay for a Dual Xeon 6 Core (12 core total), Should I
go for the AMD which will give me 4x the amount of cores or stick to Xeon,
my goal is to get the maximum amount of channels/calls  (doing lua IVR with
no transcoding) and conferencing.
Any help, feedback, benchmarks or personal experience would be appreciated
Thanks
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From gmaruzz at gmail.com  Sun Feb 14 20:28:54 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sun, 14 Feb 2016 18:28:54 +0100
Subject: [Freeswitch-users] Freeswitch doesnt transcode
In-Reply-To: <CAFoaQoAoVLu7ni7x02Y-2hS3nvjf_vdphJkLqt4+mxbhx7=g+w@mail.gmail.com>
References: <CAFoaQoAoVLu7ni7x02Y-2hS3nvjf_vdphJkLqt4+mxbhx7=g+w@mail.gmail.com>
Message-ID: <CALXCt0qKR1OusCr0i_2mrOS3i8fzY-sbu4Q_FdCuX1EBSgPA6A@mail.gmail.com>

How you originate the call? Is a bridge? From which phone?

Also, please pastebin the complete sip trace (from start of leg A to end of
both legs) and put here a link to pastebin
Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s at gmail.com> ha scritto:

> Hello,
>
> I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
> When I call to this remote sip phone i get a 406 error that opus is
> not supported as shown by the sip trace below. However, if I force the
> codec to absolute like this
> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5
> the call works fine.
>
> Is there anyway I can make FreeSWITCH to automatically transcode
> without forcing the codec string in the dial plan?
>
> The codec preferences is set as  global_codec_prefs=OPUS,PCMU,PCMA and
> outbound_codec_prefs=PCMU,PCMA,GSM
>
> ---------------------------siptrace--------------------------------
>
> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
>    ------------------------------------------------------------------------
>    SIP/2.0 406 Not Acceptable
>    Via: SIP/2.0/UDP
> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
>    Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
>    From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
>    To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
>    CSeq: 87372504 INVITE
>    Content-Length:  0
>
>    ------------------------------------------------------------------------
> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:
>    ------------------------------------------------------------------------
>    ACK sip:303 at 192.168.1.5 SIP/2.0
>    Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
>    Max-Forwards: 68
>    From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
>    To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
>    Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
>    CSeq: 87372504 ACK
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
> sofia/internal/303 at 192.168.1.5 entering state [terminated][406]
> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA]
> [SERVICE_NOT_IMPLEMENTED]
>
> Thanks
> Rajil
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 vbvbrj at gmail.com  Sun Feb 14 20:37:04 2016
From: vbvbrj at gmail.com (Mimiko)
Date: Sun, 14 Feb 2016 19:37:04 +0200
Subject: [Freeswitch-users] Fwd: hook to event or startup script which
 will hook to events?
In-Reply-To: <7ee401d1673e$93284c20$b978e460$@freeswitch.org>
References: <56B75BDE.6020604@gmail.com>
	<CAO=waFCbwKiAOnVyxSW-deT3GS7K2wGH4ZsV8H139aVUB2XpUQ@mail.gmail.com>
	<56C02E2B.2090304@gmail.com>
	<7ee401d1673e$93284c20$b978e460$@freeswitch.org>
Message-ID: <56C0BB40.3060909@gmail.com>

On 14.02.2016 17:44, Ken Rice wrote:
> Is there a ticket open on this segfault you are avoiding and how to
> reproduce it?

No, I didn't open any ticket, because my FS version is 1.3.13b 
cad607d72e compiled at 2013.01.18, so the bug may be fixed or may be 
not. I can't update right now the version.

My main question is what is best to capture event?
1) Using <hook event="RECORD_STOP" script="script.lua"/> in lua.conf.
2) or using <param name="startup-script" value="script.lua"/> in 
lua.conf and hook to event like for e in (function() return con:pop(1) 
end) do....

Comparing performance and stability.

Using 1st method has the advantage that I can change script without 
restarting FS. But at my version running it for several hours FS stops 
with segfault.

If using 1st method to capture events is better, then I'll update FS and 
see if segfault persists or not.
If 2nd method is better - then there is no need for me to update FS 
right now.

To reproduce see my first message on this.

-- 
Mimiko desu.


From vbvbrj at gmail.com  Sun Feb 14 20:39:46 2016
From: vbvbrj at gmail.com (Mimiko)
Date: Sun, 14 Feb 2016 19:39:46 +0200
Subject: [Freeswitch-users] Fwd: hook to event or startup script which
 will hook to events?
In-Reply-To: <CAOccsKB+rB8OKsRgX8YZSK0PV-sFYMs324zGGrr6V5n901=SVg@mail.gmail.com>
References: <56B75BDE.6020604@gmail.com>
	<CAO=waFCbwKiAOnVyxSW-deT3GS7K2wGH4ZsV8H139aVUB2XpUQ@mail.gmail.com>
	<56C02E2B.2090304@gmail.com>
	<CAOccsKB+rB8OKsRgX8YZSK0PV-sFYMs324zGGrr6V5n901=SVg@mail.gmail.com>
Message-ID: <56C0BBE2.5050206@gmail.com>

On 14.02.2016 16:16, ik wrote:
> I'ved created a Perl daemon for capturing specific events.
>
> You can filter to do so.
>

Do you capture all events in your perl script?

Do you need to restart perl script to capture new events after modifying 
the script? If think you do and while perl script is restarting events 
may be missed.

I prefer to use lua, because it is integrated in FS code and is much faster.

-- 
Mimiko desu.


From mike at jerris.com  Sun Feb 14 20:52:21 2016
From: mike at jerris.com (Michael Jerris)
Date: Sun, 14 Feb 2016 12:52:21 -0500
Subject: [Freeswitch-users] AMD
In-Reply-To: <CAFmZFiNO_yzv8vwO9HFM41v-B7w7mHwzBBcJ74wGmZxEm+Nq7g@mail.gmail.com>
References: <CAFmZFiNO_yzv8vwO9HFM41v-B7w7mHwzBBcJ74wGmZxEm+Nq7g@mail.gmail.com>
Message-ID: <CAEwTsAgXStGbsENve5fnYnhuVvfvGNbrVRv3fh1ieN1Q8MKwGw@mail.gmail.com>

You would have to test to see, most of my experience is a couple chip
generations back now but bang for the buck was much better with intel then
and we saw many more weird issues on AMD boxes.

On Sunday, February 14, 2016, Abaci B <abaci64 at gmail.com> wrote:

> Hi all,
> I know that in general Intel Xeon processors are recommended over AMD for
> FreeSWITCH, my question is if AMD is really bad or just not as good as
> Intel, I can get now a server with 4x AMD 6174 (48 Core total) for about
> the same price I would pay for a Dual Xeon 6 Core (12 core total), Should I
> go for the AMD which will give me 4x the amount of cores or stick to Xeon,
> my goal is to get the maximum amount of channels/calls  (doing lua IVR with
> no transcoding) and conferencing.
> Any help, feedback, benchmarks or personal experience would be appreciated
> Thanks
>
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From rajil.s at gmail.com  Sun Feb 14 22:04:32 2016
From: rajil.s at gmail.com (Rajil Saraswat)
Date: Sun, 14 Feb 2016 13:04:32 -0600
Subject: [Freeswitch-users] Freeswitch doesnt transcode
In-Reply-To: <CALXCt0qKR1OusCr0i_2mrOS3i8fzY-sbu4Q_FdCuX1EBSgPA6A@mail.gmail.com>
References: <CAFoaQoAoVLu7ni7x02Y-2hS3nvjf_vdphJkLqt4+mxbhx7=g+w@mail.gmail.com>
	<CALXCt0qKR1OusCr0i_2mrOS3i8fzY-sbu4Q_FdCuX1EBSgPA6A@mail.gmail.com>
Message-ID: <CAFoaQoASOwftNQXF5JzMvN76WwvDaf01MTz_C8CR-Y-KKVHzeA@mail.gmail.com>

The siptrace is at http://pastebin.com/xiGqtj1Y

The call is being made from 303 (Android/CSipsimple with OPUS codec)
to 208 (pjsua test client with PCMU codec). The error is on line 545.

On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> How you originate the call? Is a bridge? From which phone?
>
> Also, please pastebin the complete sip trace (from start of leg A to end of
> both legs) and put here a link to pastebin
>
> Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s at gmail.com> ha scritto:
>>
>> Hello,
>>
>> I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
>> When I call to this remote sip phone i get a 406 error that opus is
>> not supported as shown by the sip trace below. However, if I force the
>> codec to absolute like this
>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5
>> the call works fine.
>>
>> Is there anyway I can make FreeSWITCH to automatically transcode
>> without forcing the codec string in the dial plan?
>>
>> The codec preferences is set as  global_codec_prefs=OPUS,PCMU,PCMA and
>> outbound_codec_prefs=PCMU,PCMA,GSM
>>
>> ---------------------------siptrace--------------------------------
>>
>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
>>
>> ------------------------------------------------------------------------
>>    SIP/2.0 406 Not Acceptable
>>    Via: SIP/2.0/UDP
>>
>> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
>>    Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
>>    From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
>>    To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
>>    CSeq: 87372504 INVITE
>>    Content-Length:  0
>>
>>
>> ------------------------------------------------------------------------
>> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:
>>
>> ------------------------------------------------------------------------
>>    ACK sip:303 at 192.168.1.5 SIP/2.0
>>    Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
>>    Max-Forwards: 68
>>    From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
>>    To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
>>    Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
>>    CSeq: 87372504 ACK
>>    Content-Length: 0
>>
>>
>> ------------------------------------------------------------------------
>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
>> sofia/internal/303 at 192.168.1.5 entering state [terminated][406]
>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
>> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA]
>> [SERVICE_NOT_IMPLEMENTED]
>>
>> Thanks
>> Rajil
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 rtreleaven at bunnykick.ca  Sun Feb 14 22:33:49 2016
From: rtreleaven at bunnykick.ca (Russell Treleaven)
Date: Sun, 14 Feb 2016 14:33:49 -0500
Subject: [Freeswitch-users] Freeswitch doesnt transcode
In-Reply-To: <CAFoaQoASOwftNQXF5JzMvN76WwvDaf01MTz_C8CR-Y-KKVHzeA@mail.gmail.com>
References: <CAFoaQoAoVLu7ni7x02Y-2hS3nvjf_vdphJkLqt4+mxbhx7=g+w@mail.gmail.com>
	<CALXCt0qKR1OusCr0i_2mrOS3i8fzY-sbu4Q_FdCuX1EBSgPA6A@mail.gmail.com>
	<CAFoaQoASOwftNQXF5JzMvN76WwvDaf01MTz_C8CR-Y-KKVHzeA@mail.gmail.com>
Message-ID: <CAHzfiZ7zxU496F65MEGc2VD01+yg1QBa7qEt2NbN+ZjO=Ze0gw@mail.gmail.com>

Look at the invite starting at 480. You are only offering opus to the
callee.
On Feb 14, 2016 2:05 PM, "Rajil Saraswat" <rajil.s at gmail.com> wrote:

> The siptrace is at http://pastebin.com/xiGqtj1Y
>
> The call is being made from 303 (Android/CSipsimple with OPUS codec)
> to 208 (pjsua test client with PCMU codec). The error is on line 545.
>
> On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
> > How you originate the call? Is a bridge? From which phone?
> >
> > Also, please pastebin the complete sip trace (from start of leg A to end
> of
> > both legs) and put here a link to pastebin
> >
> > Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s at gmail.com> ha scritto:
> >>
> >> Hello,
> >>
> >> I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
> >> When I call to this remote sip phone i get a 406 error that opus is
> >> not supported as shown by the sip trace below. However, if I force the
> >> codec to absolute like this
> >> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5
> >> the call works fine.
> >>
> >> Is there anyway I can make FreeSWITCH to automatically transcode
> >> without forcing the codec string in the dial plan?
> >>
> >> The codec preferences is set as  global_codec_prefs=OPUS,PCMU,PCMA and
> >> outbound_codec_prefs=PCMU,PCMA,GSM
> >>
> >> ---------------------------siptrace--------------------------------
> >>
> >> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
> >>
> >> ------------------------------------------------------------------------
> >>    SIP/2.0 406 Not Acceptable
> >>    Via: SIP/2.0/UDP
> >>
> >>
> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
> >>    Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
> >>    From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
> >>    To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
> >>    CSeq: 87372504 INVITE
> >>    Content-Length:  0
> >>
> >>
> >> ------------------------------------------------------------------------
> >> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:
> >>
> >> ------------------------------------------------------------------------
> >>    ACK sip:303 at 192.168.1.5 SIP/2.0
> >>    Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
> >>    Max-Forwards: 68
> >>    From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
> >>    To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
> >>    Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
> >>    CSeq: 87372504 ACK
> >>    Content-Length: 0
> >>
> >>
> >> ------------------------------------------------------------------------
> >> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
> >> sofia/internal/303 at 192.168.1.5 entering state [terminated][406]
> >> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
> >> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA]
> >> [SERVICE_NOT_IMPLEMENTED]
> >>
> >> Thanks
> >> Rajil
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 rtreleaven at bunnykick.ca  Sun Feb 14 22:56:56 2016
From: rtreleaven at bunnykick.ca (Russell Treleaven)
Date: Sun, 14 Feb 2016 14:56:56 -0500
Subject: [Freeswitch-users] Freeswitch doesnt transcode
In-Reply-To: <CAFoaQoASOwftNQXF5JzMvN76WwvDaf01MTz_C8CR-Y-KKVHzeA@mail.gmail.com>
References: <CAFoaQoAoVLu7ni7x02Y-2hS3nvjf_vdphJkLqt4+mxbhx7=g+w@mail.gmail.com>
	<CALXCt0qKR1OusCr0i_2mrOS3i8fzY-sbu4Q_FdCuX1EBSgPA6A@mail.gmail.com>
	<CAFoaQoASOwftNQXF5JzMvN76WwvDaf01MTz_C8CR-Y-KKVHzeA@mail.gmail.com>
Message-ID: <CAHzfiZ5ckpCetwT6HYAJiH-GRJ32FgQObqCuO0LPK_wSEff74Q@mail.gmail.com>

fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation

On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <rajil.s at gmail.com> wrote:

> The siptrace is at http://pastebin.com/xiGqtj1Y
>
> The call is being made from 303 (Android/CSipsimple with OPUS codec)
> to 208 (pjsua test client with PCMU codec). The error is on line 545.
>
> On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
> > How you originate the call? Is a bridge? From which phone?
> >
> > Also, please pastebin the complete sip trace (from start of leg A to end
> of
> > both legs) and put here a link to pastebin
> >
> > Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s at gmail.com> ha scritto:
> >>
> >> Hello,
> >>
> >> I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
> >> When I call to this remote sip phone i get a 406 error that opus is
> >> not supported as shown by the sip trace below. However, if I force the
> >> codec to absolute like this
> >> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5
> >> the call works fine.
> >>
> >> Is there anyway I can make FreeSWITCH to automatically transcode
> >> without forcing the codec string in the dial plan?
> >>
> >> The codec preferences is set as  global_codec_prefs=OPUS,PCMU,PCMA and
> >> outbound_codec_prefs=PCMU,PCMA,GSM
> >>
> >> ---------------------------siptrace--------------------------------
> >>
> >> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
> >>
> >> ------------------------------------------------------------------------
> >>    SIP/2.0 406 Not Acceptable
> >>    Via: SIP/2.0/UDP
> >>
> >>
> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
> >>    Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
> >>    From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
> >>    To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
> >>    CSeq: 87372504 INVITE
> >>    Content-Length:  0
> >>
> >>
> >> ------------------------------------------------------------------------
> >> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:
> >>
> >> ------------------------------------------------------------------------
> >>    ACK sip:303 at 192.168.1.5 SIP/2.0
> >>    Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
> >>    Max-Forwards: 68
> >>    From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
> >>    To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
> >>    Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
> >>    CSeq: 87372504 ACK
> >>    Content-Length: 0
> >>
> >>
> >> ------------------------------------------------------------------------
> >> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
> >> sofia/internal/303 at 192.168.1.5 entering state [terminated][406]
> >> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
> >> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA]
> >> [SERVICE_NOT_IMPLEMENTED]
> >>
> >> Thanks
> >> Rajil
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 rajil.s at gmail.com  Mon Feb 15 00:37:47 2016
From: rajil.s at gmail.com (Rajil Saraswat)
Date: Sun, 14 Feb 2016 15:37:47 -0600
Subject: [Freeswitch-users] Freeswitch doesnt transcode
In-Reply-To: <CAHzfiZ5ckpCetwT6HYAJiH-GRJ32FgQObqCuO0LPK_wSEff74Q@mail.gmail.com>
References: <CAFoaQoAoVLu7ni7x02Y-2hS3nvjf_vdphJkLqt4+mxbhx7=g+w@mail.gmail.com>
	<CALXCt0qKR1OusCr0i_2mrOS3i8fzY-sbu4Q_FdCuX1EBSgPA6A@mail.gmail.com>
	<CAFoaQoASOwftNQXF5JzMvN76WwvDaf01MTz_C8CR-Y-KKVHzeA@mail.gmail.com>
	<CAHzfiZ5ckpCetwT6HYAJiH-GRJ32FgQObqCuO0LPK_wSEff74Q@mail.gmail.com>
Message-ID: <CAFoaQoCkvqRUT4R2LFBFKMLs9YVjp5US_kekr1EKTHUO_Z99oA@mail.gmail.com>

Thanks, after setting media_mix_inbound_outbound_codecs=true,
transcoding happens automatically.  I remember not setting this
variable in other installations and transcoding used to work out of
the box. Is media_mix_inbound_outbound_codecs=true default in
Freeswitch?

On 14 February 2016 at 13:56, Russell Treleaven <rtreleaven at bunnykick.ca> wrote:
> fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation
>
> On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <rajil.s at gmail.com> wrote:
>>
>> The siptrace is at http://pastebin.com/xiGqtj1Y
>>
>> The call is being made from 303 (Android/CSipsimple with OPUS codec)
>> to 208 (pjsua test client with PCMU codec). The error is on line 545.
>>
>> On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz at gmail.com>
>> wrote:
>> > How you originate the call? Is a bridge? From which phone?
>> >
>> > Also, please pastebin the complete sip trace (from start of leg A to end
>> > of
>> > both legs) and put here a link to pastebin
>> >
>> > Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s at gmail.com> ha scritto:
>> >>
>> >> Hello,
>> >>
>> >> I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
>> >> When I call to this remote sip phone i get a 406 error that opus is
>> >> not supported as shown by the sip trace below. However, if I force the
>> >> codec to absolute like this
>> >> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5
>> >> the call works fine.
>> >>
>> >> Is there anyway I can make FreeSWITCH to automatically transcode
>> >> without forcing the codec string in the dial plan?
>> >>
>> >> The codec preferences is set as  global_codec_prefs=OPUS,PCMU,PCMA and
>> >> outbound_codec_prefs=PCMU,PCMA,GSM
>> >>
>> >> ---------------------------siptrace--------------------------------
>> >>
>> >> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
>> >>
>> >>
>> >> ------------------------------------------------------------------------
>> >>    SIP/2.0 406 Not Acceptable
>> >>    Via: SIP/2.0/UDP
>> >>
>> >>
>> >> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
>> >>    Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
>> >>    From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
>> >>    To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
>> >>    CSeq: 87372504 INVITE
>> >>    Content-Length:  0
>> >>
>> >>
>> >>
>> >> ------------------------------------------------------------------------
>> >> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:
>> >>
>> >>
>> >> ------------------------------------------------------------------------
>> >>    ACK sip:303 at 192.168.1.5 SIP/2.0
>> >>    Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
>> >>    Max-Forwards: 68
>> >>    From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
>> >>    To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
>> >>    Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
>> >>    CSeq: 87372504 ACK
>> >>    Content-Length: 0
>> >>
>> >>
>> >>
>> >> ------------------------------------------------------------------------
>> >> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
>> >> sofia/internal/303 at 192.168.1.5 entering state [terminated][406]
>> >> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
>> >> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA]
>> >> [SERVICE_NOT_IMPLEMENTED]
>> >>
>> >> Thanks
>> >> Rajil
>> >>
>> >>
>> >> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://confluence.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Mon Feb 15 01:07:14 2016
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 14 Feb 2016 16:07:14 -0600
Subject: [Freeswitch-users] Freeswitch doesnt transcode
In-Reply-To: <CAFoaQoCkvqRUT4R2LFBFKMLs9YVjp5US_kekr1EKTHUO_Z99oA@mail.gmail.com>
References: <CAFoaQoAoVLu7ni7x02Y-2hS3nvjf_vdphJkLqt4+mxbhx7=g+w@mail.gmail.com>
	<CALXCt0qKR1OusCr0i_2mrOS3i8fzY-sbu4Q_FdCuX1EBSgPA6A@mail.gmail.com>
	<CAFoaQoASOwftNQXF5JzMvN76WwvDaf01MTz_C8CR-Y-KKVHzeA@mail.gmail.com>
	<CAHzfiZ5ckpCetwT6HYAJiH-GRJ32FgQObqCuO0LPK_wSEff74Q@mail.gmail.com>
	<CAFoaQoCkvqRUT4R2LFBFKMLs9YVjp5US_kekr1EKTHUO_Z99oA@mail.gmail.com>
Message-ID: <87D80ECA-87A9-4824-B9E7-4A1EE91AB4DC@freeswitch.org>

This behavior changed a while ago. This was dictates by ever growing SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal with fragmentation and everyone refuses to fully implement sip over tcp for some reason even tho a ton of things support it and the RFCs require it

Sent from my iPhone

> On Feb 14, 2016, at 3:37 PM, Rajil Saraswat <rajil.s at gmail.com> wrote:
> 
> Thanks, after setting media_mix_inbound_outbound_codecs=true,
> transcoding happens automatically.  I remember not setting this
> variable in other installations and transcoding used to work out of
> the box. Is media_mix_inbound_outbound_codecs=true default in
> Freeswitch?
> 
>> On 14 February 2016 at 13:56, Russell Treleaven <rtreleaven at bunnykick.ca> wrote:
>> fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation
>> 
>>> On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <rajil.s at gmail.com> wrote:
>>> 
>>> The siptrace is at http://pastebin.com/xiGqtj1Y
>>> 
>>> The call is being made from 303 (Android/CSipsimple with OPUS codec)
>>> to 208 (pjsua test client with PCMU codec). The error is on line 545.
>>> 
>>> On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz at gmail.com>
>>> wrote:
>>>> How you originate the call? Is a bridge? From which phone?
>>>> 
>>>> Also, please pastebin the complete sip trace (from start of leg A to end
>>>> of
>>>> both legs) and put here a link to pastebin
>>>> 
>>>> Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s at gmail.com> ha scritto:
>>>>> 
>>>>> Hello,
>>>>> 
>>>>> I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
>>>>> When I call to this remote sip phone i get a 406 error that opus is
>>>>> not supported as shown by the sip trace below. However, if I force the
>>>>> codec to absolute like this
>>>>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5
>>>>> the call works fine.
>>>>> 
>>>>> Is there anyway I can make FreeSWITCH to automatically transcode
>>>>> without forcing the codec string in the dial plan?
>>>>> 
>>>>> The codec preferences is set as  global_codec_prefs=OPUS,PCMU,PCMA and
>>>>> outbound_codec_prefs=PCMU,PCMA,GSM
>>>>> 
>>>>> ---------------------------siptrace--------------------------------
>>>>> 
>>>>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
>>>>> 
>>>>> 
>>>>> ------------------------------------------------------------------------
>>>>>   SIP/2.0 406 Not Acceptable
>>>>>   Via: SIP/2.0/UDP
>>>>> 
>>>>> 
>>>>> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
>>>>>   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
>>>>>   From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
>>>>>   To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
>>>>>   CSeq: 87372504 INVITE
>>>>>   Content-Length:  0
>>>>> 
>>>>> 
>>>>> 
>>>>> ------------------------------------------------------------------------
>>>>> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:
>>>>> 
>>>>> 
>>>>> ------------------------------------------------------------------------
>>>>>   ACK sip:303 at 192.168.1.5 SIP/2.0
>>>>>   Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
>>>>>   Max-Forwards: 68
>>>>>   From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
>>>>>   To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
>>>>>   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
>>>>>   CSeq: 87372504 ACK
>>>>>   Content-Length: 0
>>>>> 
>>>>> 
>>>>> 
>>>>> ------------------------------------------------------------------------
>>>>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
>>>>> sofia/internal/303 at 192.168.1.5 entering state [terminated][406]
>>>>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
>>>>> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA]
>>>>> [SERVICE_NOT_IMPLEMENTED]
>>>>> 
>>>>> Thanks
>>>>> Rajil
>>>>> 
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> 
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 colton.conor at gmail.com  Mon Feb 15 03:20:39 2016
From: colton.conor at gmail.com (Colton Conor)
Date: Sun, 14 Feb 2016 18:20:39 -0600
Subject: [Freeswitch-users] Freeswitch doesnt transcode
In-Reply-To: <87D80ECA-87A9-4824-B9E7-4A1EE91AB4DC@freeswitch.org>
References: <CAFoaQoAoVLu7ni7x02Y-2hS3nvjf_vdphJkLqt4+mxbhx7=g+w@mail.gmail.com>
	<CALXCt0qKR1OusCr0i_2mrOS3i8fzY-sbu4Q_FdCuX1EBSgPA6A@mail.gmail.com>
	<CAFoaQoASOwftNQXF5JzMvN76WwvDaf01MTz_C8CR-Y-KKVHzeA@mail.gmail.com>
	<CAHzfiZ5ckpCetwT6HYAJiH-GRJ32FgQObqCuO0LPK_wSEff74Q@mail.gmail.com>
	<CAFoaQoCkvqRUT4R2LFBFKMLs9YVjp5US_kekr1EKTHUO_Z99oA@mail.gmail.com>
	<87D80ECA-87A9-4824-B9E7-4A1EE91AB4DC@freeswitch.org>
Message-ID: <CAMDdSzNiDFhza8pFO=Udmi+0ptgu8uM8rbLXjubo0qFkDiGykw@mail.gmail.com>

So is TCP the preferred method of doing SIP these days? I like TCP with
endpoints as they always break through firewalls and we never seem to have
in issue with TCP. However UDP is a headache. So if you have the choice why
not do TCP? I realize some devices only support UDP, but the majority of
SIP phones out there today do support TCP.

Plus if you use TLS for encryption and security then you are already using
TCP right?

On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice <krice at freeswitch.org> wrote:

> This behavior changed a while ago. This was dictates by ever growing SDPs
> and exceeding MTUs causing udp fragmentation. Udp does not deal with
> fragmentation and everyone refuses to fully implement sip over tcp for some
> reason even tho a ton of things support it and the RFCs require it
>
> Sent from my iPhone
>
> > On Feb 14, 2016, at 3:37 PM, Rajil Saraswat <rajil.s at gmail.com> wrote:
> >
> > Thanks, after setting media_mix_inbound_outbound_codecs=true,
> > transcoding happens automatically.  I remember not setting this
> > variable in other installations and transcoding used to work out of
> > the box. Is media_mix_inbound_outbound_codecs=true default in
> > Freeswitch?
> >
> >> On 14 February 2016 at 13:56, Russell Treleaven <
> rtreleaven at bunnykick.ca> wrote:
> >> fyi
> https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation
> >>
> >>> On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <rajil.s at gmail.com>
> wrote:
> >>>
> >>> The siptrace is at http://pastebin.com/xiGqtj1Y
> >>>
> >>> The call is being made from 303 (Android/CSipsimple with OPUS codec)
> >>> to 208 (pjsua test client with PCMU codec). The error is on line 545.
> >>>
> >>> On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz at gmail.com>
> >>> wrote:
> >>>> How you originate the call? Is a bridge? From which phone?
> >>>>
> >>>> Also, please pastebin the complete sip trace (from start of leg A to
> end
> >>>> of
> >>>> both legs) and put here a link to pastebin
> >>>>
> >>>> Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s at gmail.com> ha
> scritto:
> >>>>>
> >>>>> Hello,
> >>>>>
> >>>>> I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
> >>>>> When I call to this remote sip phone i get a 406 error that opus is
> >>>>> not supported as shown by the sip trace below. However, if I force
> the
> >>>>> codec to absolute like this
> >>>>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5
> >>>>> the call works fine.
> >>>>>
> >>>>> Is there anyway I can make FreeSWITCH to automatically transcode
> >>>>> without forcing the codec string in the dial plan?
> >>>>>
> >>>>> The codec preferences is set as  global_codec_prefs=OPUS,PCMU,PCMA
> and
> >>>>> outbound_codec_prefs=PCMU,PCMA,GSM
> >>>>>
> >>>>> ---------------------------siptrace--------------------------------
> >>>>>
> >>>>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
> >>>>>
> >>>>>
> >>>>>
> ------------------------------------------------------------------------
> >>>>>   SIP/2.0 406 Not Acceptable
> >>>>>   Via: SIP/2.0/UDP
> >>>>>
> >>>>>
> >>>>>
> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
> >>>>>   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
> >>>>>   From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
> >>>>>   To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
> >>>>>   CSeq: 87372504 INVITE
> >>>>>   Content-Length:  0
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> ------------------------------------------------------------------------
> >>>>> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:
> >>>>>
> >>>>>
> >>>>>
> ------------------------------------------------------------------------
> >>>>>   ACK sip:303 at 192.168.1.5 SIP/2.0
> >>>>>   Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
> >>>>>   Max-Forwards: 68
> >>>>>   From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
> >>>>>   To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
> >>>>>   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
> >>>>>   CSeq: 87372504 ACK
> >>>>>   Content-Length: 0
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> ------------------------------------------------------------------------
> >>>>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
> >>>>> sofia/internal/303 at 192.168.1.5 entering state [terminated][406]
> >>>>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
> >>>>> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA]
> >>>>> [SERVICE_NOT_IMPLEMENTED]
> >>>>>
> >>>>> Thanks
> >>>>> Rajil
> >>>>>
> >>>>>
> >>>>>
> _________________________________________________________________________
> >>>>> Professional FreeSWITCH Consulting Services:
> >>>>> consulting at freeswitch.org
> >>>>> http://www.freeswitchsolutions.com
> >>>>>
> >>>>> Official FreeSWITCH Sites
> >>>>> http://www.freeswitch.org
> >>>>> http://confluence.freeswitch.org
> >>>>> http://www.cluecon.com
> >>>>>
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>
> >>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>
> >>>>
> >>>>
> >>>>
> _________________________________________________________________________
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>>
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://confluence.freeswitch.org
> >>>> http://www.cluecon.com
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://confluence.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 idokan at gmail.com  Mon Feb 15 10:41:12 2016
From: idokan at gmail.com (ik)
Date: Mon, 15 Feb 2016 09:41:12 +0200
Subject: [Freeswitch-users] Fwd: hook to event or startup script which
 will hook to events?
In-Reply-To: <56C0BBE2.5050206@gmail.com>
References: <56B75BDE.6020604@gmail.com>
	<CAO=waFCbwKiAOnVyxSW-deT3GS7K2wGH4ZsV8H139aVUB2XpUQ@mail.gmail.com>
	<56C02E2B.2090304@gmail.com>
	<CAOccsKB+rB8OKsRgX8YZSK0PV-sFYMs324zGGrr6V5n901=SVg@mail.gmail.com>
	<56C0BBE2.5050206@gmail.com>
Message-ID: <CAOccsKB4xTVAdBCF2VAbFrqtmjPOjxT83mv3wnxwd8iZ_JxEzw@mail.gmail.com>

On my case, I listen only for specific events, so I use the filter api
call, but without it, you will get all events.
On Feb 14, 2016 7:41 PM, "Mimiko" <vbvbrj at gmail.com> wrote:

> On 14.02.2016 16:16, ik wrote:
> > I'ved created a Perl daemon for capturing specific events.
> >
> > You can filter to do so.
> >
>
> Do you capture all events in your perl script?
>
> Do you need to restart perl script to capture new events after modifying
> the script? If think you do and while perl script is restarting events
> may be missed.
>
> I prefer to use lua, because it is integrated in FS code and is much
> faster.
>
> --
> Mimiko desu.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From stefano.favaro at edistar.com  Mon Feb 15 12:12:25 2016
From: stefano.favaro at edistar.com (Stefano Favaro)
Date: Mon, 15 Feb 2016 10:12:25 +0100 (CET)
Subject: [Freeswitch-users] Problem with mod_spy
In-Reply-To: <CABMX_w-SYNKF4_E5Ui+ou0y-qvM8awG0nodCvTgKqRZRtbJ=cw@mail.gmail.com>
Message-ID: <1597675782.6819.1455527545027.JavaMail.root@mailserver.edistar.com>

Thank you Italo. 

This is my log 

https://pastebin.freeswitch.org/24555 

and this is the id of my user spy call 

94bae5a0-dd2c-4d0e-99fd-8ef81e753227 

Stefano 

----- Messaggio originale -----
Da: "?talo Rossi" <italo at freeswitch.org> 
A: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org> 
Inviato: Gioved?, 11 febbraio 2016 18:52:46 
Oggetto: Re: [Freeswitch-users] Problem with mod_spy 


Stefano, 


Can you post your debug logs (/log 7)? 


Use https://pastebin.freeswitch.org/ 


On Thu, Feb 11, 2016 at 1:33 PM, Stefano Favaro < stefano.favaro at edistar.com > wrote: 




Hello,

 I have a problem with the mod_spy module. 
It seems that it just plays music and do not actually spy.
I want to dial an extension with the number of the user I want to spy and wait for a call in or out for that user. 
Mod_spy application plays music but when a call is handled by the user I can't hear it continues to play the music on hold audio file. 

This is the dialplan:

<extension name="eavesdrop"> 
<condition field="destination_number" expression="^88(.*)$|^\*0(.*)$"> 
<action application="answer"/> 
<action application="userspy" data="$1 at myserver"/> 
</condition> 
</extension> 
If I dial 881000, for example, It means I want to spy on user 1000.
I have in and out calls from user 1000 but I can't hear. 
userspy_show in fs_cli, I get : 
1000 at myserver : e9165eaf-e8d7-40ad-a8cd-d2963f363684

1 total spy 

I have FreeSwitch version FreeSWITCH Version 1.4.26-37~64bit (-37 64bit) 

SF. 

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

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

FreeSWITCH-users mailing list 
FreeSWITCH-users at 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 matt at supportedbusiness.com  Mon Feb 15 17:28:11 2016
From: matt at supportedbusiness.com (Matt Broad)
Date: Mon, 15 Feb 2016 14:28:11 +0000
Subject: [Freeswitch-users] bind_digit_action in a conference
Message-ID: <CAEeV61jqj6Hj+HQposUHuODh8APnY8Sv9LBdmmPRpyG=mq-juA@mail.gmail.com>

Hi,

is it possible to bind a digit action and set the target as another member
of a conference?

I would like to be able to press a digit and have another member execute an
extension where they enter some digits before returning back to the
conference.

thanks
Matt
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From adnan.ahmed1 at gmail.com  Mon Feb 15 03:10:02 2016
From: adnan.ahmed1 at gmail.com (Adnan Ahmed)
Date: Mon, 15 Feb 2016 00:10:02 +0000
Subject: [Freeswitch-users] FS Bridged call, no RTP until DTMF pressed
In-Reply-To: <CAORkSaavPyN8hhd6hEu6oJS4tUHo=Ug+vzAqq=gy-r2CbrC97A@mail.gmail.com>
References: <CAORkSaavPyN8hhd6hEu6oJS4tUHo=Ug+vzAqq=gy-r2CbrC97A@mail.gmail.com>
Message-ID: <CAORkSaacCWc8VCLczDsOjftiC+2TiiX0QnsvS9MgecdK-QqQmQ@mail.gmail.com>

You can disregard this message. Further testing and isolation of the
software/hardware found the cause to be in the chan_dahdi.c file for
asterisk.  (Not actually an error but it was what was causing me issues.)

The conf mute was being turned on because it thought it was receiving a
DTMFdown event on the E&m wink with Feature Group B MF signalling trunk.
Not sure if it was an error or anything, so I just hard coded the file to
not use confmute, and now it works fine.

Adnan.

On Thu, Feb 11, 2016, 09:32 Adnan Ahmed <adnan.ahmed1 at gmail.com> wrote:

> Hi,
>
> I have a peculiar situation in which I'm hoping someone can help me out
> with.  I have a Dahdi trunk coming into Asterisk (*), which then sends the
> call directly to freeswitch (FS), FS will then bridge this incoming call to
> a SIP device.  The problem i'm having is that when FS bridges the call
> there is no media (or RTP packets) sent back to asterisk until I press a
> dtmf key from the caller side.
>
> The reason that * is there is due to the fact that mod_freeTDM for FS
> wasn't able to configure the trunk parameters required to control the T1
> (E&M with Feature Group B MF), with chan_dahdi in * i was able to set that
> up with signalling=featb.  The dialplan in asterisk is as follows,
>
> [from-pstn]
>> exten => _X.,1,NoOp(Incoming DID matches as ${EXTEN})
>> exten => _X.,n,Answer()
>> exten => _X.,n,Set(CALLERID(all)="0000000000"<0000000000>)
>> exten => _X.,n,Dial(SIP/freeswitch/1819${EXTEN:0:7},90,M(send-dtmf-1)r)
>> exten => _X.,n,Hangup()
>>
>
>
>> [macro-send-dtmf-1]
>> exten => s,1,SendDTMF(1)
>
>
>  I tried sending a DTMF from astersk, and FS recognizes the DTMF, but
> still no RTP until the key is physically pressed on the caller side.  The
> asterisk dialplan is very simple, answer the incoming dahdi call and send
> it to FS via SIP.  Once the DTMF is pressed, the audio is complete and no
> issues anymore, so its not a routing, or firewall issue.  Both asterisk and
> FS run on the same machine (* on port 5065, and FS on 5060).  Looking at
> the tcpdump traces, there really is no RTP from FS until after the DTMF is
> pressed, but the RTP from asterisk is always there.
>
> I have the output of "sofia global siptrace on" at the following pastebin:
> https://pastebin.freeswitch.org/24552
>
> In that SIP trace you will see the call as follows,
>
> Incoming call from *
> bridge to SIP device
> Failure to connect to SIP device
> Forward call to voicemail
> bridge to voicemail
> connects to voicemail system
> hangup
>
> I can press the DTMF at any point once the first bridge is dialed and will
> start hearing the audio from that point onwards ... in this case i pressed
> the DTMF key 1 (you see it being recognized in the FS sip trace log).  It
> makes no difference if I wait to press the DTMF till the second bridge or
> after the second bridge connects.
>
> I have even tried it with a sip device that answers on the first bridge
> session, and its the same scenario:  no audio until dtmf is pressed, again
> making no difference if its pressed right away or 10 seconds after the call
> is connected and the other party can hear me but i don't hear them until i
> press the dtmf.
>
> Thanks,
> Adnan.
>
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From silviu.cpp at gmail.com  Mon Feb 15 17:34:52 2016
From: silviu.cpp at gmail.com (Caragea Silviu)
Date: Mon, 15 Feb 2016 16:34:52 +0200
Subject: [Freeswitch-users] freeswitch 1.6 on Ubuntu 14.04 LTS
Message-ID: <CAJsc=ovuAGfmj=TF4jyNEx7=f5sEYv36Wd1uTx=7mwfV=8VyEw@mail.gmail.com>

Hello,

Is Ubuntu 14.04 supported on FS 1.6 ? Seems there are compilation issues
for several modules

Silviu
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From gmaruzz at gmail.com  Mon Feb 15 18:22:37 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 15 Feb 2016 16:22:37 +0100
Subject: [Freeswitch-users] freeswitch 1.6 on Ubuntu 14.04 LTS
In-Reply-To: <CAJsc=ovuAGfmj=TF4jyNEx7=f5sEYv36Wd1uTx=7mwfV=8VyEw@mail.gmail.com>
References: <CAJsc=ovuAGfmj=TF4jyNEx7=f5sEYv36Wd1uTx=7mwfV=8VyEw@mail.gmail.com>
Message-ID: <CALXCt0oWbaEK8BJFG10qcxTF4eVKv3kcTm54NTd-8mjaGxMVYA@mail.gmail.com>

At this moment in time, Debian 8 (Jessie) is the recommended and supported
platform.
Il 15/Feb/2016 15:36, "Caragea Silviu" <silviu.cpp at gmail.com> ha scritto:

> Hello,
>
> Is Ubuntu 14.04 supported on FS 1.6 ? Seems there are compilation issues
> for several modules
>
> 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 mike at jerris.com  Mon Feb 15 18:33:46 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 15 Feb 2016 10:33:46 -0500
Subject: [Freeswitch-users] Freeswitch doesnt transcode
In-Reply-To: <CAMDdSzNiDFhza8pFO=Udmi+0ptgu8uM8rbLXjubo0qFkDiGykw@mail.gmail.com>
References: <CAFoaQoAoVLu7ni7x02Y-2hS3nvjf_vdphJkLqt4+mxbhx7=g+w@mail.gmail.com>
	<CALXCt0qKR1OusCr0i_2mrOS3i8fzY-sbu4Q_FdCuX1EBSgPA6A@mail.gmail.com>
	<CAFoaQoASOwftNQXF5JzMvN76WwvDaf01MTz_C8CR-Y-KKVHzeA@mail.gmail.com>
	<CAHzfiZ5ckpCetwT6HYAJiH-GRJ32FgQObqCuO0LPK_wSEff74Q@mail.gmail.com>
	<CAFoaQoCkvqRUT4R2LFBFKMLs9YVjp5US_kekr1EKTHUO_Z99oA@mail.gmail.com>
	<87D80ECA-87A9-4824-B9E7-4A1EE91AB4DC@freeswitch.org>
	<CAMDdSzNiDFhza8pFO=Udmi+0ptgu8uM8rbLXjubo0qFkDiGykw@mail.gmail.com>
Message-ID: <CAEwTsAj7jy7uPzESmZamhNZF76uVFBp8YJ63sU_2jFyH7vayHQ@mail.gmail.com>

any device that even remotely follows sip
specs supports TCP.  Most phones I have seen do.

On Sunday, February 14, 2016, Colton Conor <colton.conor at gmail.com> wrote:

> So is TCP the preferred method of doing SIP these days? I like TCP with
> endpoints as they always break through firewalls and we never seem to have
> in issue with TCP. However UDP is a headache. So if you have the choice why
> not do TCP? I realize some devices only support UDP, but the majority of
> SIP phones out there today do support TCP.
>
> Plus if you use TLS for encryption and security then you are already using
> TCP right?
>
> On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice <krice at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','krice at freeswitch.org');>> wrote:
>
>> This behavior changed a while ago. This was dictates by ever growing SDPs
>> and exceeding MTUs causing udp fragmentation. Udp does not deal with
>> fragmentation and everyone refuses to fully implement sip over tcp for some
>> reason even tho a ton of things support it and the RFCs require it
>>
>> Sent from my iPhone
>>
>> > On Feb 14, 2016, at 3:37 PM, Rajil Saraswat <rajil.s at gmail.com
>> <javascript:_e(%7B%7D,'cvml','rajil.s at gmail.com');>> wrote:
>> >
>> > Thanks, after setting media_mix_inbound_outbound_codecs=true,
>> > transcoding happens automatically.  I remember not setting this
>> > variable in other installations and transcoding used to work out of
>> > the box. Is media_mix_inbound_outbound_codecs=true default in
>> > Freeswitch?
>> >
>> >> On 14 February 2016 at 13:56, Russell Treleaven <
>> rtreleaven at bunnykick.ca
>> <javascript:_e(%7B%7D,'cvml','rtreleaven at bunnykick.ca');>> wrote:
>> >> fyi
>> https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation
>> >>
>> >>> On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <rajil.s at gmail.com
>> <javascript:_e(%7B%7D,'cvml','rajil.s at gmail.com');>> wrote:
>> >>>
>> >>> The siptrace is at http://pastebin.com/xiGqtj1Y
>> >>>
>> >>> The call is being made from 303 (Android/CSipsimple with OPUS codec)
>> >>> to 208 (pjsua test client with PCMU codec). The error is on line 545.
>> >>>
>> >>> On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz at gmail.com
>> <javascript:_e(%7B%7D,'cvml','gmaruzz at gmail.com');>>
>> >>> wrote:
>> >>>> How you originate the call? Is a bridge? From which phone?
>> >>>>
>> >>>> Also, please pastebin the complete sip trace (from start of leg A to
>> end
>> >>>> of
>> >>>> both legs) and put here a link to pastebin
>> >>>>
>> >>>> Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s at gmail.com
>> <javascript:_e(%7B%7D,'cvml','rajil.s at gmail.com');>> ha scritto:
>> >>>>>
>> >>>>> Hello,
>> >>>>>
>> >>>>> I have a remote sip phone (Linksys SPA3102) which only supports
>> PCMU.
>> >>>>> When I call to this remote sip phone i get a 406 error that opus is
>> >>>>> not supported as shown by the sip trace below. However, if I force
>> the
>> >>>>> codec to absolute like this
>> >>>>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5
>> <javascript:_e(%7B%7D,'cvml','303 at 192.168.1.5');>
>> >>>>> the call works fine.
>> >>>>>
>> >>>>> Is there anyway I can make FreeSWITCH to automatically transcode
>> >>>>> without forcing the codec string in the dial plan?
>> >>>>>
>> >>>>> The codec preferences is set as  global_codec_prefs=OPUS,PCMU,PCMA
>> and
>> >>>>> outbound_codec_prefs=PCMU,PCMA,GSM
>> >>>>>
>> >>>>> ---------------------------siptrace--------------------------------
>> >>>>>
>> >>>>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
>> >>>>>
>> >>>>>
>> >>>>>
>> ------------------------------------------------------------------------
>> >>>>>   SIP/2.0 406 Not Acceptable
>> >>>>>   Via: SIP/2.0/UDP
>> >>>>>
>> >>>>>
>> >>>>>
>> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
>> >>>>>   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
>> >>>>>   From: "202" <sip:202 at 192.168.1.111
>> <javascript:_e(%7B%7D,'cvml','sip:202 at 192.168.1.111');>
>> >;tag=DFX0FUvr2vNcm
>> >>>>>   To: <sip:303 at 192.168.1.5
>> <javascript:_e(%7B%7D,'cvml','sip:303 at 192.168.1.5');>
>> >;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
>> >>>>>   CSeq: 87372504 INVITE
>> >>>>>   Content-Length:  0
>> >>>>>
>> >>>>>
>> >>>>>
>> >>>>>
>> ------------------------------------------------------------------------
>> >>>>> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:
>> >>>>>
>> >>>>>
>> >>>>>
>> ------------------------------------------------------------------------
>> >>>>>   ACK sip:303 at 192.168.1.5
>> <javascript:_e(%7B%7D,'cvml','sip:303 at 192.168.1.5');> SIP/2.0
>> >>>>>   Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
>> >>>>>   Max-Forwards: 68
>> >>>>>   From: "202" <sip:202 at 192.168.1.111
>> <javascript:_e(%7B%7D,'cvml','sip:202 at 192.168.1.111');>
>> >;tag=DFX0FUvr2vNcm
>> >>>>>   To: <sip:303 at 192.168.1.5
>> <javascript:_e(%7B%7D,'cvml','sip:303 at 192.168.1.5');>
>> >;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
>> >>>>>   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
>> >>>>>   CSeq: 87372504 ACK
>> >>>>>   Content-Length: 0
>> >>>>>
>> >>>>>
>> >>>>>
>> >>>>>
>> ------------------------------------------------------------------------
>> >>>>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
>> >>>>> sofia/internal/303 at 192.168.1.5
>> <javascript:_e(%7B%7D,'cvml','303 at 192.168.1.5');> entering state
>> [terminated][406]
>> >>>>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
>> >>>>> sofia/internal/303 at 192.168.1.5
>> <javascript:_e(%7B%7D,'cvml','303 at 192.168.1.5');> [CS_CONSUME_MEDIA]
>> >>>>> [SERVICE_NOT_IMPLEMENTED]
>> >>>>>
>> >>>>> Thanks
>> >>>>> Rajil
>> >>>>>
>> >>>>>
>> >>>>>
>> _________________________________________________________________________
>> >>>>> 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
>>
>> _________________________________________________________________________
>> 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 Feb 15 18:35:31 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 15 Feb 2016 10:35:31 -0500
Subject: [Freeswitch-users] bind_digit_action in a conference
In-Reply-To: <CAEeV61jqj6Hj+HQposUHuODh8APnY8Sv9LBdmmPRpyG=mq-juA@mail.gmail.com>
References: <CAEeV61jqj6Hj+HQposUHuODh8APnY8Sv9LBdmmPRpyG=mq-juA@mail.gmail.com>
Message-ID: <CAEwTsAjFfKAnwC6nRCowBPtizznPbJYA9HX-SY9SRDQ=YVKoWg@mail.gmail.com>

maybe if you entered a lua script and did those more advanced actions it
could be done

On Monday, February 15, 2016, Matt Broad <matt at supportedbusiness.com> wrote:

> Hi,
>
> is it possible to bind a digit action and set the target as another member
> of a conference?
>
> I would like to be able to press a digit and have another member execute
> an extension where they enter some digits before returning back to the
> conference.
>
> thanks
> Matt
>
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From mitch.capper at gmail.com  Mon Feb 15 18:51:02 2016
From: mitch.capper at gmail.com (Mitch Capper)
Date: Mon, 15 Feb 2016 07:51:02 -0800
Subject: [Freeswitch-users] freeswitch 1.6 on Ubuntu 14.04 LTS
In-Reply-To: <CAJsc=ovuAGfmj=TF4jyNEx7=f5sEYv36Wd1uTx=7mwfV=8VyEw@mail.gmail.com>
References: <CAJsc=ovuAGfmj=TF4jyNEx7=f5sEYv36Wd1uTx=7mwfV=8VyEw@mail.gmail.com>
Message-ID: <CALn233pSah+OPFr7MfdMbEk=skh3qvc9=-1giCq5C+bH4cD3QQ@mail.gmail.com>

I haven't had an issue with 14 or 15 with FS with the stock modules (and a
few extras I use).

~mitch

On Mon, Feb 15, 2016 at 6:34 AM, Caragea Silviu <silviu.cpp at gmail.com>
wrote:

> Hello,
>
> Is Ubuntu 14.04 supported on FS 1.6 ? Seems there are compilation issues
> for several modules
>
> 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 matt at supportedbusiness.com  Mon Feb 15 19:00:56 2016
From: matt at supportedbusiness.com (Matt Broad)
Date: Mon, 15 Feb 2016 16:00:56 +0000
Subject: [Freeswitch-users] bind_digit_action in a conference
In-Reply-To: <CAEwTsAjFfKAnwC6nRCowBPtizznPbJYA9HX-SY9SRDQ=YVKoWg@mail.gmail.com>
References: <CAEeV61jqj6Hj+HQposUHuODh8APnY8Sv9LBdmmPRpyG=mq-juA@mail.gmail.com>
	<CAEwTsAjFfKAnwC6nRCowBPtizznPbJYA9HX-SY9SRDQ=YVKoWg@mail.gmail.com>
Message-ID: <CAEeV61ggm3wbDumPHnvK4WCvkd84eSBivZwyk=T49mwN17HgcA@mail.gmail.com>

Hi Michael,

thanks for the response.  I have tried this using javascript, but the
digits are not collected.

A quick example is I sit in a loop until the mod presses the relevant key,
on the keypress I clone the session of the user who is to enter the digits
and have that execute an extension.

while (callConnected == true)
{
  if(getDigits == true)
  {
      //set back to false to prevent this code block from being hit on the
next loop
      getDigits = false;
      var _session = new session(uuid);
      _session.execute("execute_extension", "collectdigits XML
Testfeatures");
  }
}

<extension name="collectdigits ">
        <condition field="destination_number" expression="^collectdigits $">
            <action application="flush_dtmf"/>
            <action application="play_and_get_digits" data="3 3 2 5000 #
$${sound_prefix}/reception/EnterDigits.mp3
$${sound_prefix}//${audiobase}tryagain.mp3 myDigits" />
        </condition>
    </extension>


If I set the bind_digit_action to the session that will enter the digits,
it works fine but the issue is I would like the other member of the
conference to instigate the request for digits

_session.execute("bind_digit_action",
"card_digits,6,exec:execute_extension,collectdigits XML
Testfeatures,self,self");


thanks
Matt

On 15 February 2016 at 15:35, Michael Jerris <mike at jerris.com> wrote:

> maybe if you entered a lua script and did those more advanced actions it
> could be done
>
>
> On Monday, February 15, 2016, Matt Broad <matt at supportedbusiness.com>
> wrote:
>
>> Hi,
>>
>> is it possible to bind a digit action and set the target as another
>> member of a conference?
>>
>> I would like to be able to press a digit and have another member execute
>> an extension where they enter some digits before returning back to the
>> conference.
>>
>> 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 brian at freeswitch.org  Mon Feb 15 19:03:01 2016
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Feb 2016 10:03:01 -0600
Subject: [Freeswitch-users] freeswitch 1.6 on Ubuntu 14.04 LTS
In-Reply-To: <CALn233pSah+OPFr7MfdMbEk=skh3qvc9=-1giCq5C+bH4cD3QQ@mail.gmail.com>
References: <CAJsc=ovuAGfmj=TF4jyNEx7=f5sEYv36Wd1uTx=7mwfV=8VyEw@mail.gmail.com>
	<CALn233pSah+OPFr7MfdMbEk=skh3qvc9=-1giCq5C+bH4cD3QQ@mail.gmail.com>
Message-ID: <CAEJMVkB1jrdgE5uj-FnCp6H963VTAhsBOvv0Gko_474C6oVeRQ@mail.gmail.com>

For those interested https://www.gofundme.com/freeswitch_ubuntu

We are working on Official packages, if you care to donate to the cause.

/b

On Mon, Feb 15, 2016 at 9:51 AM, Mitch Capper <mitch.capper at gmail.com>
wrote:

> I haven't had an issue with 14 or 15 with FS with the stock modules (and a
> few extras I use).
>
> ~mitch
>
> On Mon, Feb 15, 2016 at 6:34 AM, Caragea Silviu <silviu.cpp at gmail.com>
> wrote:
>
>> Hello,
>>
>> Is Ubuntu 14.04 supported on FS 1.6 ? Seems there are compilation issues
>> for several modules
>>
>> 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
>



-- 

*Brian West*
brian at freeswitch.org


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

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 colton.conor at gmail.com  Mon Feb 15 19:04:13 2016
From: colton.conor at gmail.com (Colton Conor)
Date: Mon, 15 Feb 2016 10:04:13 -0600
Subject: [Freeswitch-users] Freeswitch doesnt transcode
In-Reply-To: <CAEwTsAj7jy7uPzESmZamhNZF76uVFBp8YJ63sU_2jFyH7vayHQ@mail.gmail.com>
References: <CAFoaQoAoVLu7ni7x02Y-2hS3nvjf_vdphJkLqt4+mxbhx7=g+w@mail.gmail.com>
	<CALXCt0qKR1OusCr0i_2mrOS3i8fzY-sbu4Q_FdCuX1EBSgPA6A@mail.gmail.com>
	<CAFoaQoASOwftNQXF5JzMvN76WwvDaf01MTz_C8CR-Y-KKVHzeA@mail.gmail.com>
	<CAHzfiZ5ckpCetwT6HYAJiH-GRJ32FgQObqCuO0LPK_wSEff74Q@mail.gmail.com>
	<CAFoaQoCkvqRUT4R2LFBFKMLs9YVjp5US_kekr1EKTHUO_Z99oA@mail.gmail.com>
	<87D80ECA-87A9-4824-B9E7-4A1EE91AB4DC@freeswitch.org>
	<CAMDdSzNiDFhza8pFO=Udmi+0ptgu8uM8rbLXjubo0qFkDiGykw@mail.gmail.com>
	<CAEwTsAj7jy7uPzESmZamhNZF76uVFBp8YJ63sU_2jFyH7vayHQ@mail.gmail.com>
Message-ID: <CAMDdSzM3aVHTwi5DAGgfdqFEt8FO60YPfhPDYQUyrsyVYP2CLQ@mail.gmail.com>

So if the device supports TCP, is there any reason not to use TCP. AKA is
there any reason to keep on using UDP. TCP seems superior.

On Mon, Feb 15, 2016 at 9:33 AM, Michael Jerris <mike at jerris.com> wrote:

> any device that even remotely follows sip
> specs supports TCP.  Most phones I have seen do.
>
>
> On Sunday, February 14, 2016, Colton Conor <colton.conor at gmail.com> wrote:
>
>> So is TCP the preferred method of doing SIP these days? I like TCP with
>> endpoints as they always break through firewalls and we never seem to have
>> in issue with TCP. However UDP is a headache. So if you have the choice why
>> not do TCP? I realize some devices only support UDP, but the majority of
>> SIP phones out there today do support TCP.
>>
>> Plus if you use TLS for encryption and security then you are already
>> using TCP right?
>>
>> On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice <krice at freeswitch.org> wrote:
>>
>>> This behavior changed a while ago. This was dictates by ever growing
>>> SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal with
>>> fragmentation and everyone refuses to fully implement sip over tcp for some
>>> reason even tho a ton of things support it and the RFCs require it
>>>
>>> Sent from my iPhone
>>>
>>> > On Feb 14, 2016, at 3:37 PM, Rajil Saraswat <rajil.s at gmail.com> wrote:
>>> >
>>> > Thanks, after setting media_mix_inbound_outbound_codecs=true,
>>> > transcoding happens automatically.  I remember not setting this
>>> > variable in other installations and transcoding used to work out of
>>> > the box. Is media_mix_inbound_outbound_codecs=true default in
>>> > Freeswitch?
>>> >
>>> >> On 14 February 2016 at 13:56, Russell Treleaven <
>>> rtreleaven at bunnykick.ca> wrote:
>>> >> fyi
>>> https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation
>>> >>
>>> >>> On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <rajil.s at gmail.com>
>>> wrote:
>>> >>>
>>> >>> The siptrace is at http://pastebin.com/xiGqtj1Y
>>> >>>
>>> >>> The call is being made from 303 (Android/CSipsimple with OPUS codec)
>>> >>> to 208 (pjsua test client with PCMU codec). The error is on line 545.
>>> >>>
>>> >>> On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz at gmail.com
>>> >
>>> >>> wrote:
>>> >>>> How you originate the call? Is a bridge? From which phone?
>>> >>>>
>>> >>>> Also, please pastebin the complete sip trace (from start of leg A
>>> to end
>>> >>>> of
>>> >>>> both legs) and put here a link to pastebin
>>> >>>>
>>> >>>> Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s at gmail.com> ha
>>> scritto:
>>> >>>>>
>>> >>>>> Hello,
>>> >>>>>
>>> >>>>> I have a remote sip phone (Linksys SPA3102) which only supports
>>> PCMU.
>>> >>>>> When I call to this remote sip phone i get a 406 error that opus is
>>> >>>>> not supported as shown by the sip trace below. However, if I force
>>> the
>>> >>>>> codec to absolute like this
>>> >>>>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5
>>> >>>>> the call works fine.
>>> >>>>>
>>> >>>>> Is there anyway I can make FreeSWITCH to automatically transcode
>>> >>>>> without forcing the codec string in the dial plan?
>>> >>>>>
>>> >>>>> The codec preferences is set as  global_codec_prefs=OPUS,PCMU,PCMA
>>> and
>>> >>>>> outbound_codec_prefs=PCMU,PCMA,GSM
>>> >>>>>
>>> >>>>> ---------------------------siptrace--------------------------------
>>> >>>>>
>>> >>>>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
>>> >>>>>
>>> >>>>>
>>> >>>>>
>>> ------------------------------------------------------------------------
>>> >>>>>   SIP/2.0 406 Not Acceptable
>>> >>>>>   Via: SIP/2.0/UDP
>>> >>>>>
>>> >>>>>
>>> >>>>>
>>> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
>>> >>>>>   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
>>> >>>>>   From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
>>> >>>>>   To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
>>> >>>>>   CSeq: 87372504 INVITE
>>> >>>>>   Content-Length:  0
>>> >>>>>
>>> >>>>>
>>> >>>>>
>>> >>>>>
>>> ------------------------------------------------------------------------
>>> >>>>> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:
>>> >>>>>
>>> >>>>>
>>> >>>>>
>>> ------------------------------------------------------------------------
>>> >>>>>   ACK sip:303 at 192.168.1.5 SIP/2.0
>>> >>>>>   Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
>>> >>>>>   Max-Forwards: 68
>>> >>>>>   From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
>>> >>>>>   To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
>>> >>>>>   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
>>> >>>>>   CSeq: 87372504 ACK
>>> >>>>>   Content-Length: 0
>>> >>>>>
>>> >>>>>
>>> >>>>>
>>> >>>>>
>>> ------------------------------------------------------------------------
>>> >>>>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
>>> >>>>> sofia/internal/303 at 192.168.1.5 entering state [terminated][406]
>>> >>>>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
>>> >>>>> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA]
>>> >>>>> [SERVICE_NOT_IMPLEMENTED]
>>> >>>>>
>>> >>>>> Thanks
>>> >>>>> Rajil
>>> >>>>>
>>> >>>>>
>>> >>>>>
>>> _________________________________________________________________________
>>> >>>>> Professional FreeSWITCH Consulting Services:
>>> >>>>> consulting at freeswitch.org
>>> >>>>> http://www.freeswitchsolutions.com
>>> >>>>>
>>> >>>>> Official FreeSWITCH Sites
>>> >>>>> http://www.freeswitch.org
>>> >>>>> http://confluence.freeswitch.org
>>> >>>>> http://www.cluecon.com
>>> >>>>>
>>> >>>>> FreeSWITCH-users mailing list
>>> >>>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>>>
>>> >>>>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>>> http://www.freeswitch.org
>>> >>>>
>>> >>>>
>>> >>>>
>>> >>>>
>>> _________________________________________________________________________
>>> >>>> Professional FreeSWITCH Consulting Services:
>>> >>>> consulting at freeswitch.org
>>> >>>> http://www.freeswitchsolutions.com
>>> >>>>
>>> >>>> Official FreeSWITCH Sites
>>> >>>> http://www.freeswitch.org
>>> >>>> http://confluence.freeswitch.org
>>> >>>> http://www.cluecon.com
>>> >>>>
>>> >>>> FreeSWITCH-users mailing list
>>> >>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>> http://www.freeswitch.org
>>> >>>
>>> >>>
>>> _________________________________________________________________________
>>> >>> Professional FreeSWITCH Consulting Services:
>>> >>> consulting at freeswitch.org
>>> >>> http://www.freeswitchsolutions.com
>>> >>>
>>> >>> Official FreeSWITCH Sites
>>> >>> http://www.freeswitch.org
>>> >>> http://confluence.freeswitch.org
>>> >>> http://www.cluecon.com
>>> >>>
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>
>>> >>
>>> >>
>>> >>
>>> _________________________________________________________________________
>>> >> Professional FreeSWITCH Consulting Services:
>>> >> consulting at freeswitch.org
>>> >> http://www.freeswitchsolutions.com
>>> >>
>>> >> Official FreeSWITCH Sites
>>> >> http://www.freeswitch.org
>>> >> http://confluence.freeswitch.org
>>> >> http://www.cluecon.com
>>> >>
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://confluence.freeswitch.org
>>> > http://www.cluecon.com
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb 15 19:08:52 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 15 Feb 2016 11:08:52 -0500
Subject: [Freeswitch-users] Freeswitch doesnt transcode
In-Reply-To: <CAMDdSzM3aVHTwi5DAGgfdqFEt8FO60YPfhPDYQUyrsyVYP2CLQ@mail.gmail.com>
References: <CAFoaQoAoVLu7ni7x02Y-2hS3nvjf_vdphJkLqt4+mxbhx7=g+w@mail.gmail.com>
	<CALXCt0qKR1OusCr0i_2mrOS3i8fzY-sbu4Q_FdCuX1EBSgPA6A@mail.gmail.com>
	<CAFoaQoASOwftNQXF5JzMvN76WwvDaf01MTz_C8CR-Y-KKVHzeA@mail.gmail.com>
	<CAHzfiZ5ckpCetwT6HYAJiH-GRJ32FgQObqCuO0LPK_wSEff74Q@mail.gmail.com>
	<CAFoaQoCkvqRUT4R2LFBFKMLs9YVjp5US_kekr1EKTHUO_Z99oA@mail.gmail.com>
	<87D80ECA-87A9-4824-B9E7-4A1EE91AB4DC@freeswitch.org>
	<CAMDdSzNiDFhza8pFO=Udmi+0ptgu8uM8rbLXjubo0qFkDiGykw@mail.gmail.com>
	<CAEwTsAj7jy7uPzESmZamhNZF76uVFBp8YJ63sU_2jFyH7vayHQ@mail.gmail.com>
	<CAMDdSzM3aVHTwi5DAGgfdqFEt8FO60YPfhPDYQUyrsyVYP2CLQ@mail.gmail.com>
Message-ID: <8402F14B-6417-4A62-98BF-C8C4CA07D425@jerris.com>

It is heavier but I think that otherwise is superior.

> On Feb 15, 2016, at 11:04 AM, Colton Conor <colton.conor at gmail.com> wrote:
> 
> So if the device supports TCP, is there any reason not to use TCP. AKA is there any reason to keep on using UDP. TCP seems superior. 
> 
> On Mon, Feb 15, 2016 at 9:33 AM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> any device that even remotely follows sip
> specs supports TCP.  Most phones I have seen do.
> 
> 
> On Sunday, February 14, 2016, Colton Conor <colton.conor at gmail.com <mailto:colton.conor at gmail.com>> wrote:
> So is TCP the preferred method of doing SIP these days? I like TCP with endpoints as they always break through firewalls and we never seem to have in issue with TCP. However UDP is a headache. So if you have the choice why not do TCP? I realize some devices only support UDP, but the majority of SIP phones out there today do support TCP.
> 
> Plus if you use TLS for encryption and security then you are already using TCP right? 
> 
> On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice <krice at freeswitch.org <>> wrote:
> This behavior changed a while ago. This was dictates by ever growing SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal with fragmentation and everyone refuses to fully implement sip over tcp for some reason even tho a ton of things support it and the RFCs require it
> 
> Sent from my iPhone
> 
> > On Feb 14, 2016, at 3:37 PM, Rajil Saraswat <rajil.s at gmail.com <>> wrote:
> >
> > Thanks, after setting media_mix_inbound_outbound_codecs=true,
> > transcoding happens automatically.  I remember not setting this
> > variable in other installations and transcoding used to work out of
> > the box. Is media_mix_inbound_outbound_codecs=true default in
> > Freeswitch?
> >
> >> On 14 February 2016 at 13:56, Russell Treleaven <rtreleaven at bunnykick.ca <>> wrote:
> >> fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation <https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation>
> >>
> >>> On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <rajil.s at gmail.com <>> wrote:
> >>>
> >>> The siptrace is at http://pastebin.com/xiGqtj1Y <http://pastebin.com/xiGqtj1Y>
> >>>
> >>> The call is being made from 303 (Android/CSipsimple with OPUS codec)
> >>> to 208 (pjsua test client with PCMU codec). The error is on line 545.
> >>>
> >>> On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz at gmail.com <>>
> >>> wrote:
> >>>> How you originate the call? Is a bridge? From which phone?
> >>>>
> >>>> Also, please pastebin the complete sip trace (from start of leg A to end
> >>>> of
> >>>> both legs) and put here a link to pastebin
> >>>>
> >>>> Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s at gmail.com <>> ha scritto:
> >>>>>
> >>>>> Hello,
> >>>>>
> >>>>> I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
> >>>>> When I call to this remote sip phone i get a 406 error that opus is
> >>>>> not supported as shown by the sip trace below. However, if I force the
> >>>>> codec to absolute like this
> >>>>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5 <>
> >>>>> the call works fine.
> >>>>>
> >>>>> Is there anyway I can make FreeSWITCH to automatically transcode
> >>>>> without forcing the codec string in the dial plan?
> >>>>>
> >>>>> The codec preferences is set as  global_codec_prefs=OPUS,PCMU,PCMA and
> >>>>> outbound_codec_prefs=PCMU,PCMA,GSM
> >>>>>
> >>>>> ---------------------------siptrace--------------------------------
> >>>>>
> >>>>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
> >>>>>
> >>>>>
> >>>>> ------------------------------------------------------------------------
> >>>>>   SIP/2.0 406 Not Acceptable
> >>>>>   Via: SIP/2.0/UDP
> >>>>>
> >>>>>
> >>>>> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
> >>>>>   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
> >>>>>   From: "202" <sip:202 at 192.168.1.111 <>>;tag=DFX0FUvr2vNcm
> >>>>>   To: <sip:303 at 192.168.1.5 <>>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
> >>>>>   CSeq: 87372504 INVITE
> >>>>>   Content-Length:  0
> >>>>>
> >>>>>
> >>>>>
> >>>>> ------------------------------------------------------------------------
> >>>>> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:
> >>>>>
> >>>>>
> >>>>> ------------------------------------------------------------------------
> >>>>>   ACK sip:303 at 192.168.1.5 <> SIP/2.0
> >>>>>   Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
> >>>>>   Max-Forwards: 68
> >>>>>   From: "202" <sip:202 at 192.168.1.111 <>>;tag=DFX0FUvr2vNcm
> >>>>>   To: <sip:303 at 192.168.1.5 <>>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
> >>>>>   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
> >>>>>   CSeq: 87372504 ACK
> >>>>>   Content-Length: 0
> >>>>>
> >>>>>
> >>>>>
> >>>>> ------------------------------------------------------------------------
> >>>>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
> >>>>> sofia/internal/303 at 192.168.1.5 <> entering state [terminated][406]
> >>>>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
> >>>>> sofia/internal/303 at 192.168.1.5 <> [CS_CONSUME_MEDIA]
> >>>>> [SERVICE_NOT_IMPLEMENTED]
> >>>>>
> >>>>> Thanks
> >>>>> Rajil
> >>>>>
> >>>>>
> >>>>> _________________________________________________________________________
> >>>>> Professional FreeSWITCH Consulting Services:
> >>>>> 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 <>
> >>>>> 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 <http://www.freeswitchsolutions.com/>
> >>>>
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org <http://www.freeswitch.org/>
> >>>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> >>>> http://www.cluecon.com <http://www.cluecon.com/>
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org <>
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> >>>> 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 <http://www.freeswitchsolutions.com/>
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org <http://www.freeswitch.org/>
> >>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> >>> http://www.cluecon.com <http://www.cluecon.com/>
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org <>
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> >>> 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 <http://www.freeswitchsolutions.com/>
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org <http://www.freeswitch.org/>
> >> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> >> http://www.cluecon.com <http://www.cluecon.com/>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org <>
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> >> 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 <http://www.freeswitchsolutions.com/>
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org <http://www.freeswitch.org/>
> > http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> > http://www.cluecon.com <http://www.cluecon.com/>
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org <>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> > 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 <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> 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 mandra at gmail.com  Mon Feb 15 19:16:36 2016
From: mandra at gmail.com (Chris Mandra)
Date: Mon, 15 Feb 2016 11:16:36 -0500
Subject: [Freeswitch-users] bridging out and custom signaling question
Message-ID: <CAJdRzN5fJzDV-e7o6_sj4cy_Z-D8W+EG=qKpkkg33PyL0+7Udw@mail.gmail.com>

Hi guys - I hope things are good where you are. I have a question:

What is the easiest way to bridge a call out of freeswitch not using sip
but using a custom signaling protocol that may be a restful protocol?

Thank you !

chris
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From krice at freeswitch.org  Mon Feb 15 19:25:11 2016
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 15 Feb 2016 10:25:11 -0600
Subject: [Freeswitch-users] Freeswitch doesnt transcode
In-Reply-To: <CAMDdSzM3aVHTwi5DAGgfdqFEt8FO60YPfhPDYQUyrsyVYP2CLQ@mail.gmail.com>
References: <CAFoaQoAoVLu7ni7x02Y-2hS3nvjf_vdphJkLqt4+mxbhx7=g+w@mail.gmail.com>	<CALXCt0qKR1OusCr0i_2mrOS3i8fzY-sbu4Q_FdCuX1EBSgPA6A@mail.gmail.com>	<CAFoaQoASOwftNQXF5JzMvN76WwvDaf01MTz_C8CR-Y-KKVHzeA@mail.gmail.com>	<CAHzfiZ5ckpCetwT6HYAJiH-GRJ32FgQObqCuO0LPK_wSEff74Q@mail.gmail.com>	<CAFoaQoCkvqRUT4R2LFBFKMLs9YVjp5US_kekr1EKTHUO_Z99oA@mail.gmail.com>	<87D80ECA-87A9-4824-B9E7-4A1EE91AB4DC@freeswitch.org>	<CAMDdSzNiDFhza8pFO=Udmi+0ptgu8uM8rbLXjubo0qFkDiGykw@mail.gmail.com>	<CAEwTsAj7jy7uPzESmZamhNZF76uVFBp8YJ63sU_2jFyH7vayHQ@mail.gmail.com>
	<CAMDdSzM3aVHTwi5DAGgfdqFEt8FO60YPfhPDYQUyrsyVYP2CLQ@mail.gmail.com>
Message-ID: <109501d1680d$723c40f0$56b4c2d0$@freeswitch.org>

The problem isn?t necessarily the devices, but there is also the carriers?  

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Colton Conor
Sent: Monday, February 15, 2016 10:04 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode

 

So if the device supports TCP, is there any reason not to use TCP. AKA is there any reason to keep on using UDP. TCP seems superior. 

 

On Mon, Feb 15, 2016 at 9:33 AM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com> > wrote:

any device that even remotely follows sip

specs supports TCP.  Most phones I have seen do.



On Sunday, February 14, 2016, Colton Conor <colton.conor at gmail.com <mailto:colton.conor at gmail.com> > wrote:

So is TCP the preferred method of doing SIP these days? I like TCP with endpoints as they always break through firewalls and we never seem to have in issue with TCP. However UDP is a headache. So if you have the choice why not do TCP? I realize some devices only support UDP, but the majority of SIP phones out there today do support TCP.

 

Plus if you use TLS for encryption and security then you are already using TCP right? 

 

On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org> > wrote:

This behavior changed a while ago. This was dictates by ever growing SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal with fragmentation and everyone refuses to fully implement sip over tcp for some reason even tho a ton of things support it and the RFCs require it

Sent from my iPhone


> On Feb 14, 2016, at 3:37 PM, Rajil Saraswat <rajil.s at gmail.com <mailto:rajil.s at gmail.com> > wrote:
>
> Thanks, after setting media_mix_inbound_outbound_codecs=true,
> transcoding happens automatically.  I remember not setting this
> variable in other installations and transcoding used to work out of
> the box. Is media_mix_inbound_outbound_codecs=true default in
> Freeswitch?
>
>> On 14 February 2016 at 13:56, Russell Treleaven <rtreleaven at bunnykick.ca <mailto:rtreleaven at bunnykick.ca> > wrote:
>> fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation
>>
>>> On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <rajil.s at gmail.com <mailto:rajil.s at gmail.com> > wrote:
>>>
>>> The siptrace is at http://pastebin.com/xiGqtj1Y
>>>
>>> The call is being made from 303 (Android/CSipsimple with OPUS codec)
>>> to 208 (pjsua test client with PCMU codec). The error is on line 545.
>>>
>>> On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com> >
>>> wrote:
>>>> How you originate the call? Is a bridge? From which phone?
>>>>
>>>> Also, please pastebin the complete sip trace (from start of leg A to end
>>>> of
>>>> both legs) and put here a link to pastebin
>>>>
>>>> Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s at gmail.com <mailto:rajil.s at gmail.com> > ha scritto:
>>>>>
>>>>> Hello,
>>>>>
>>>>> I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
>>>>> When I call to this remote sip phone i get a 406 error that opus is
>>>>> not supported as shown by the sip trace below. However, if I force the
>>>>> codec to absolute like this
>>>>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5
>>>>> the call works fine.
>>>>>
>>>>> Is there anyway I can make FreeSWITCH to automatically transcode
>>>>> without forcing the codec string in the dial plan?
>>>>>
>>>>> The codec preferences is set as  global_codec_prefs=OPUS,PCMU,PCMA and
>>>>> outbound_codec_prefs=PCMU,PCMA,GSM
>>>>>
>>>>> ---------------------------siptrace--------------------------------
>>>>>
>>>>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>   SIP/2.0 406 Not Acceptable
>>>>>   Via: SIP/2.0/UDP
>>>>>
>>>>>
>>>>> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
>>>>>   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
>>>>>   From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
>>>>>   To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
>>>>>   CSeq: 87372504 INVITE
>>>>>   Content-Length:  0
>>>>>
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>   ACK sip:303 at 192.168.1.5 SIP/2.0
>>>>>   Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
>>>>>   Max-Forwards: 68
>>>>>   From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
>>>>>   To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
>>>>>   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
>>>>>   CSeq: 87372504 ACK
>>>>>   Content-Length: 0
>>>>>
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
>>>>> sofia/internal/303 at 192.168.1.5 <mailto:sofia/internal/303 at 192.168.1.5>  entering state [terminated][406]
>>>>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
>>>>> sofia/internal/303 at 192.168.1.5 <mailto:sofia/internal/303 at 192.168.1.5>  [CS_CONSUME_MEDIA]
>>>>> [SERVICE_NOT_IMPLEMENTED]
>>>>>
>>>>> Thanks
>>>>> Rajil
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> 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 <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 <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 colton.conor at gmail.com  Mon Feb 15 19:29:40 2016
From: colton.conor at gmail.com (Colton Conor)
Date: Mon, 15 Feb 2016 10:29:40 -0600
Subject: [Freeswitch-users] Freeswitch doesnt transcode
In-Reply-To: <109501d1680d$723c40f0$56b4c2d0$@freeswitch.org>
References: <CAFoaQoAoVLu7ni7x02Y-2hS3nvjf_vdphJkLqt4+mxbhx7=g+w@mail.gmail.com>
	<CALXCt0qKR1OusCr0i_2mrOS3i8fzY-sbu4Q_FdCuX1EBSgPA6A@mail.gmail.com>
	<CAFoaQoASOwftNQXF5JzMvN76WwvDaf01MTz_C8CR-Y-KKVHzeA@mail.gmail.com>
	<CAHzfiZ5ckpCetwT6HYAJiH-GRJ32FgQObqCuO0LPK_wSEff74Q@mail.gmail.com>
	<CAFoaQoCkvqRUT4R2LFBFKMLs9YVjp5US_kekr1EKTHUO_Z99oA@mail.gmail.com>
	<87D80ECA-87A9-4824-B9E7-4A1EE91AB4DC@freeswitch.org>
	<CAMDdSzNiDFhza8pFO=Udmi+0ptgu8uM8rbLXjubo0qFkDiGykw@mail.gmail.com>
	<CAEwTsAj7jy7uPzESmZamhNZF76uVFBp8YJ63sU_2jFyH7vayHQ@mail.gmail.com>
	<CAMDdSzM3aVHTwi5DAGgfdqFEt8FO60YPfhPDYQUyrsyVYP2CLQ@mail.gmail.com>
	<109501d1680d$723c40f0$56b4c2d0$@freeswitch.org>
Message-ID: <CAMDdSzMh_fsQ+pUNGu3SkrJ3U8BOVvjXnpaJ7GtK5PanmHbH8A@mail.gmail.com>

True,

But freeswitch talking to the carriers is almost always UPD.

However, freeswitch talking to the clients I would say TCP would be idea.
So its almost like freeswitch is trancoding from TCP to UDP too :)

On Mon, Feb 15, 2016 at 10:25 AM, Ken Rice <krice at freeswitch.org> wrote:

> The problem isn?t necessarily the devices, but there is also the carriers?
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Colton Conor
> *Sent:* Monday, February 15, 2016 10:04 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Freeswitch doesnt transcode
>
>
>
> So if the device supports TCP, is there any reason not to use TCP. AKA is
> there any reason to keep on using UDP. TCP seems superior.
>
>
>
> On Mon, Feb 15, 2016 at 9:33 AM, Michael Jerris <mike at jerris.com> wrote:
>
> any device that even remotely follows sip
>
> specs supports TCP.  Most phones I have seen do.
>
>
>
> On Sunday, February 14, 2016, Colton Conor <colton.conor at gmail.com> wrote:
>
> So is TCP the preferred method of doing SIP these days? I like TCP with
> endpoints as they always break through firewalls and we never seem to have
> in issue with TCP. However UDP is a headache. So if you have the choice why
> not do TCP? I realize some devices only support UDP, but the majority of
> SIP phones out there today do support TCP.
>
>
>
> Plus if you use TLS for encryption and security then you are already using
> TCP right?
>
>
>
> On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice <krice at freeswitch.org> wrote:
>
> This behavior changed a while ago. This was dictates by ever growing SDPs
> and exceeding MTUs causing udp fragmentation. Udp does not deal with
> fragmentation and everyone refuses to fully implement sip over tcp for some
> reason even tho a ton of things support it and the RFCs require it
>
> Sent from my iPhone
>
>
> > On Feb 14, 2016, at 3:37 PM, Rajil Saraswat <rajil.s at gmail.com> wrote:
> >
> > Thanks, after setting media_mix_inbound_outbound_codecs=true,
> > transcoding happens automatically.  I remember not setting this
> > variable in other installations and transcoding used to work out of
> > the box. Is media_mix_inbound_outbound_codecs=true default in
> > Freeswitch?
> >
> >> On 14 February 2016 at 13:56, Russell Treleaven <
> rtreleaven at bunnykick.ca> wrote:
> >> fyi
> https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation
> >>
> >>> On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <rajil.s at gmail.com>
> wrote:
> >>>
> >>> The siptrace is at http://pastebin.com/xiGqtj1Y
> >>>
> >>> The call is being made from 303 (Android/CSipsimple with OPUS codec)
> >>> to 208 (pjsua test client with PCMU codec). The error is on line 545.
> >>>
> >>> On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz at gmail.com>
> >>> wrote:
> >>>> How you originate the call? Is a bridge? >From which phone?
> >>>>
> >>>> Also, please pastebin the complete sip trace (from start of leg A to
> end
> >>>> of
> >>>> both legs) and put here a link to pastebin
> >>>>
> >>>> Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s at gmail.com> ha
> scritto:
> >>>>>
> >>>>> Hello,
> >>>>>
> >>>>> I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
> >>>>> When I call to this remote sip phone i get a 406 error that opus is
> >>>>> not supported as shown by the sip trace below. However, if I force
> the
> >>>>> codec to absolute like this
> >>>>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5
> >>>>> the call works fine.
> >>>>>
> >>>>> Is there anyway I can make FreeSWITCH to automatically transcode
> >>>>> without forcing the codec string in the dial plan?
> >>>>>
> >>>>> The codec preferences is set as  global_codec_prefs=OPUS,PCMU,PCMA
> and
> >>>>> outbound_codec_prefs=PCMU,PCMA,GSM
> >>>>>
> >>>>> ---------------------------siptrace--------------------------------
> >>>>>
> >>>>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
> >>>>>
> >>>>>
> >>>>>
> ------------------------------------------------------------------------
> >>>>>   SIP/2.0 406 Not Acceptable
> >>>>>   Via: SIP/2.0/UDP
> >>>>>
> >>>>>
> >>>>>
> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
> >>>>>   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
> >>>>>   From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
> >>>>>   To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
> >>>>>   CSeq: 87372504 INVITE
> >>>>>   Content-Length:  0
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> ------------------------------------------------------------------------
> >>>>> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:
> >>>>>
> >>>>>
> >>>>>
> ------------------------------------------------------------------------
> >>>>>   ACK sip:303 at 192.168.1.5 SIP/2.0
> >>>>>   Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
> >>>>>   Max-Forwards: 68
> >>>>>   From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
> >>>>>   To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
> >>>>>   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
> >>>>>   CSeq: 87372504 ACK
> >>>>>   Content-Length: 0
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> ------------------------------------------------------------------------
> >>>>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
> >>>>> sofia/internal/303 at 192.168.1.5 entering state [terminated][406]
> >>>>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
> >>>>> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA]
> >>>>> [SERVICE_NOT_IMPLEMENTED]
> >>>>>
> >>>>> Thanks
> >>>>> Rajil
> >>>>>
> >>>>>
> >>>>>
> _________________________________________________________________________
> >>>>> Professional FreeSWITCH Consulting Services:
> >>>>> consulting at freeswitch.org
> >>>>> http://www.freeswitchsolutions.com
> >>>>>
> >>>>> Official FreeSWITCH Sites
> >>>>> http://www.freeswitch.org
> >>>>> http://confluence.freeswitch.org
> >>>>> http://www.cluecon.com
> >>>>>
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>
> >>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>
> >>>>
> >>>>
> >>>>
> _________________________________________________________________________
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>>
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://confluence.freeswitch.org
> >>>> http://www.cluecon.com
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://confluence.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb 15 19:35:05 2016
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 15 Feb 2016 10:35:05 -0600
Subject: [Freeswitch-users] Freeswitch doesnt transcode
In-Reply-To: <CAMDdSzMh_fsQ+pUNGu3SkrJ3U8BOVvjXnpaJ7GtK5PanmHbH8A@mail.gmail.com>
References: <CAFoaQoAoVLu7ni7x02Y-2hS3nvjf_vdphJkLqt4+mxbhx7=g+w@mail.gmail.com>	<CALXCt0qKR1OusCr0i_2mrOS3i8fzY-sbu4Q_FdCuX1EBSgPA6A@mail.gmail.com>	<CAFoaQoASOwftNQXF5JzMvN76WwvDaf01MTz_C8CR-Y-KKVHzeA@mail.gmail.com>	<CAHzfiZ5ckpCetwT6HYAJiH-GRJ32FgQObqCuO0LPK_wSEff74Q@mail.gmail.com>	<CAFoaQoCkvqRUT4R2LFBFKMLs9YVjp5US_kekr1EKTHUO_Z99oA@mail.gmail.com>	<87D80ECA-87A9-4824-B9E7-4A1EE91AB4DC@freeswitch.org>	<CAMDdSzNiDFhza8pFO=Udmi+0ptgu8uM8rbLXjubo0qFkDiGykw@mail.gmail.com>	<CAEwTsAj7jy7uPzESmZamhNZF76uVFBp8YJ63sU_2jFyH7vayHQ@mail.gmail.com>	<CAMDdSzM3aVHTwi5DAGgfdqFEt8FO60YPfhPDYQUyrsyVYP2CLQ@mail.gmail.com>	<109501d1680d$723c40f0$56b4c2d0$@freeswitch.org>
	<CAMDdSzMh_fsQ+pUNGu3SkrJ3U8BOVvjXnpaJ7GtK5PanmHbH8A@mail.gmail.com>
Message-ID: <109f01d1680e$d3baf190$7b30d4b0$@freeswitch.org>

The problem still exists for expanding SDPs? using TCP to the user/device then trying to send the same thing out to the carrier over UDP is what was causing the problem in the first place? so the decision was made to prevent those problems we?ll only offer what the device offers and not expand the number of codecs even further increasing the already bloated SDPs to the point where they fragment over UDP and get dropped? 

 

So is TCP better for some things, yes it is, however, the lack of market wide support for it with carriers makes it a pain in the ass even tho the RFCs specifically say you MUST support both UDP and TCP for SIP, but certain VoIP softwares out there only implemented UDP many years ago and now we?re stuck with that legacy

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Colton Conor
Sent: Monday, February 15, 2016 10:30 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode

 

True,

 

But freeswitch talking to the carriers is almost always UPD.

 

However, freeswitch talking to the clients I would say TCP would be idea. So its almost like freeswitch is trancoding from TCP to UDP too :)

 

On Mon, Feb 15, 2016 at 10:25 AM, Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org> > wrote:

The problem isn?t necessarily the devices, but there is also the carriers?  

 

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 Colton Conor
Sent: Monday, February 15, 2016 10:04 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode

 

So if the device supports TCP, is there any reason not to use TCP. AKA is there any reason to keep on using UDP. TCP seems superior. 

 

On Mon, Feb 15, 2016 at 9:33 AM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com> > wrote:

any device that even remotely follows sip

specs supports TCP.  Most phones I have seen do.



On Sunday, February 14, 2016, Colton Conor <colton.conor at gmail.com <mailto:colton.conor at gmail.com> > wrote:

So is TCP the preferred method of doing SIP these days? I like TCP with endpoints as they always break through firewalls and we never seem to have in issue with TCP. However UDP is a headache. So if you have the choice why not do TCP? I realize some devices only support UDP, but the majority of SIP phones out there today do support TCP.

 

Plus if you use TLS for encryption and security then you are already using TCP right? 

 

On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org> > wrote:

This behavior changed a while ago. This was dictates by ever growing SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal with fragmentation and everyone refuses to fully implement sip over tcp for some reason even tho a ton of things support it and the RFCs require it

Sent from my iPhone


> On Feb 14, 2016, at 3:37 PM, Rajil Saraswat <rajil.s at gmail.com <mailto:rajil.s at gmail.com> > wrote:
>
> Thanks, after setting media_mix_inbound_outbound_codecs=true,
> transcoding happens automatically.  I remember not setting this
> variable in other installations and transcoding used to work out of
> the box. Is media_mix_inbound_outbound_codecs=true default in
> Freeswitch?
>
>> On 14 February 2016 at 13:56, Russell Treleaven <rtreleaven at bunnykick.ca <mailto:rtreleaven at bunnykick.ca> > wrote:
>> fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation
>>
>>> On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <rajil.s at gmail.com <mailto:rajil.s at gmail.com> > wrote:
>>>
>>> The siptrace is at http://pastebin.com/xiGqtj1Y
>>>
>>> The call is being made from 303 (Android/CSipsimple with OPUS codec)
>>> to 208 (pjsua test client with PCMU codec). The error is on line 545.
>>>
>>> On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com> >
>>> wrote:
>>>> How you originate the call? Is a bridge? >From which phone?
>>>>
>>>> Also, please pastebin the complete sip trace (from start of leg A to end
>>>> of
>>>> both legs) and put here a link to pastebin
>>>>
>>>> Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s at gmail.com <mailto:rajil.s at gmail.com> > ha scritto:
>>>>>
>>>>> Hello,
>>>>>
>>>>> I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
>>>>> When I call to this remote sip phone i get a 406 error that opus is
>>>>> not supported as shown by the sip trace below. However, if I force the
>>>>> codec to absolute like this
>>>>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5 <mailto:303 at 192.168.1.5> 
>>>>> the call works fine.
>>>>>
>>>>> Is there anyway I can make FreeSWITCH to automatically transcode
>>>>> without forcing the codec string in the dial plan?
>>>>>
>>>>> The codec preferences is set as  global_codec_prefs=OPUS,PCMU,PCMA and
>>>>> outbound_codec_prefs=PCMU,PCMA,GSM
>>>>>
>>>>> ---------------------------siptrace--------------------------------
>>>>>
>>>>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>   SIP/2.0 406 Not Acceptable
>>>>>   Via: SIP/2.0/UDP
>>>>>
>>>>>
>>>>> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
>>>>>   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
>>>>>   From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
>>>>>   To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
>>>>>   CSeq: 87372504 INVITE
>>>>>   Content-Length:  0
>>>>>
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>   ACK sip:303 at 192.168.1.5 SIP/2.0
>>>>>   Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
>>>>>   Max-Forwards: 68
>>>>>   From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
>>>>>   To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
>>>>>   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
>>>>>   CSeq: 87372504 ACK
>>>>>   Content-Length: 0
>>>>>
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
>>>>> sofia/internal/303 at 192.168.1.5 <mailto:sofia/internal/303 at 192.168.1.5>  entering state [terminated][406]
>>>>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
>>>>> sofia/internal/303 at 192.168.1.5 <mailto:sofia/internal/303 at 192.168.1.5>  [CS_CONSUME_MEDIA]
>>>>> [SERVICE_NOT_IMPLEMENTED]
>>>>>
>>>>> Thanks
>>>>> Rajil
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> 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 <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 <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 jalsot at gmail.com  Mon Feb 15 20:39:06 2016
From: jalsot at gmail.com (Tamas Jalsovszky)
Date: Mon, 15 Feb 2016 18:39:06 +0100
Subject: [Freeswitch-users] freeswitch 1.6 on Ubuntu 14.04 LTS
In-Reply-To: <CAEJMVkB1jrdgE5uj-FnCp6H963VTAhsBOvv0Gko_474C6oVeRQ@mail.gmail.com>
References: <CAJsc=ovuAGfmj=TF4jyNEx7=f5sEYv36Wd1uTx=7mwfV=8VyEw@mail.gmail.com>
	<CALn233pSah+OPFr7MfdMbEk=skh3qvc9=-1giCq5C+bH4cD3QQ@mail.gmail.com>
	<CAEJMVkB1jrdgE5uj-FnCp6H963VTAhsBOvv0Gko_474C6oVeRQ@mail.gmail.com>
Message-ID: <CAHCLHGsyMokNKjqv3D_XVj0Lte+a-Aoa5moX9mCzibNQHfRpXg@mail.gmail.com>

Are you going to support upcoming 16.04 LTS too?

Jalsot

On Mon, Feb 15, 2016 at 5:03 PM, Brian West <brian at freeswitch.org> wrote:

> For those interested https://www.gofundme.com/freeswitch_ubuntu
>
> We are working on Official packages, if you care to donate to the cause.
>
> /b
>
> On Mon, Feb 15, 2016 at 9:51 AM, Mitch Capper <mitch.capper at gmail.com>
> wrote:
>
>> I haven't had an issue with 14 or 15 with FS with the stock modules (and
>> a few extras I use).
>>
>> ~mitch
>>
>> On Mon, Feb 15, 2016 at 6:34 AM, Caragea Silviu <silviu.cpp at gmail.com>
>> wrote:
>>
>>> Hello,
>>>
>>> Is Ubuntu 14.04 supported on FS 1.6 ? Seems there are compilation issues
>>> for several modules
>>>
>>> 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
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> 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 brian at freeswitch.org  Mon Feb 15 20:54:33 2016
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Feb 2016 11:54:33 -0600
Subject: [Freeswitch-users] freeswitch 1.6 on Ubuntu 14.04 LTS
In-Reply-To: <CAHCLHGsyMokNKjqv3D_XVj0Lte+a-Aoa5moX9mCzibNQHfRpXg@mail.gmail.com>
References: <CAJsc=ovuAGfmj=TF4jyNEx7=f5sEYv36Wd1uTx=7mwfV=8VyEw@mail.gmail.com>
	<CALn233pSah+OPFr7MfdMbEk=skh3qvc9=-1giCq5C+bH4cD3QQ@mail.gmail.com>
	<CAEJMVkB1jrdgE5uj-FnCp6H963VTAhsBOvv0Gko_474C6oVeRQ@mail.gmail.com>
	<CAHCLHGsyMokNKjqv3D_XVj0Lte+a-Aoa5moX9mCzibNQHfRpXg@mail.gmail.com>
Message-ID: <CAEJMVkAW1k18gkLv9JF6dpX6nkW5Bd0OiJM8Wab_BMcem4bqVw@mail.gmail.com>

I don't see why not, it may require more testing and some work....

On Mon, Feb 15, 2016 at 11:39 AM, Tamas Jalsovszky <jalsot at gmail.com> wrote:

> Are you going to support upcoming 16.04 LTS too?
>
> Jalsot
>
> On Mon, Feb 15, 2016 at 5:03 PM, Brian West <brian at freeswitch.org> wrote:
>
>> For those interested https://www.gofundme.com/freeswitch_ubuntu
>>
>> We are working on Official packages, if you care to donate to the cause.
>>
>> /b
>>
>> On Mon, Feb 15, 2016 at 9:51 AM, Mitch Capper <mitch.capper at gmail.com>
>> wrote:
>>
>>> I haven't had an issue with 14 or 15 with FS with the stock modules (and
>>> a few extras I use).
>>>
>>> ~mitch
>>>
>>> On Mon, Feb 15, 2016 at 6:34 AM, Caragea Silviu <silviu.cpp at gmail.com>
>>> wrote:
>>>
>>>> Hello,
>>>>
>>>> Is Ubuntu 14.04 supported on FS 1.6 ? Seems there are compilation
>>>> issues for several modules
>>>>
>>>> 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
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> 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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

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 bobjectsfreeswitch at gmail.com  Mon Feb 15 21:27:26 2016
From: bobjectsfreeswitch at gmail.com (Bob Hartwig)
Date: Mon, 15 Feb 2016 12:27:26 -0600
Subject: [Freeswitch-users] Freeswitch doesnt transcode
In-Reply-To: <CAHzfiZ5ckpCetwT6HYAJiH-GRJ32FgQObqCuO0LPK_wSEff74Q@mail.gmail.com>
References: <CAFoaQoAoVLu7ni7x02Y-2hS3nvjf_vdphJkLqt4+mxbhx7=g+w@mail.gmail.com>
	<CALXCt0qKR1OusCr0i_2mrOS3i8fzY-sbu4Q_FdCuX1EBSgPA6A@mail.gmail.com>
	<CAFoaQoASOwftNQXF5JzMvN76WwvDaf01MTz_C8CR-Y-KKVHzeA@mail.gmail.com>
	<CAHzfiZ5ckpCetwT6HYAJiH-GRJ32FgQObqCuO0LPK_wSEff74Q@mail.gmail.com>
Message-ID: <CAAemr3y16uriPOdVcErFvQtupMawK67-Nc_BCNsJ3o4LK1x9FA@mail.gmail.com>

On Sun, Feb 14, 2016 at 1:56 PM, Russell Treleaven <rtreleaven at bunnykick.ca>
wrote:

> fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation
>
>
That is a really lucid and complete description of CODEC negotiation in
FS.  Thanks for the link, and thanks to the original author for writing it.

Bob
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From lconroy at insensate.co.uk  Mon Feb 15 21:44:38 2016
From: lconroy at insensate.co.uk (Lawrence Conroy)
Date: Mon, 15 Feb 2016 18:44:38 +0000
Subject: [Freeswitch-users] TCP vs UDP (was Re: Freeswitch doesnt
	transcode)
In-Reply-To: <109f01d1680e$d3baf190$7b30d4b0$@freeswitch.org>
References: <CAFoaQoAoVLu7ni7x02Y-2hS3nvjf_vdphJkLqt4+mxbhx7=g+w@mail.gmail.com>	<CALXCt0qKR1OusCr0i_2mrOS3i8fzY-sbu4Q_FdCuX1EBSgPA6A@mail.gmail.com>	<CAFoaQoASOwftNQXF5JzMvN76WwvDaf01MTz_C8CR-Y-KKVHzeA@mail.gmail.com>	<CAHzfiZ5ckpCetwT6HYAJiH-GRJ32FgQObqCuO0LPK_wSEff74Q@mail.gmail.com>	<CAFoaQoCkvqRUT4R2LFBFKMLs9YVjp5US_kekr1EKTHUO_Z99oA@mail.gmail.com>	<87D80ECA-87A9-4824-B9E7-4A1EE91AB4DC@freeswitch.org>	<CAMDdSzNiDFhza8pFO=Udmi+0ptgu8uM8rbLXjubo0qFkDiGykw@mail.gmail.com>	<CAEwTsAj7jy7uPzESmZamhNZF76uVFBp8YJ63sU_2jFyH7vayHQ@mail.gmail.com>	<CAMDdSzM3aVHTwi5DAGgfdqFEt8FO60YPfhPDYQUyrsyVYP2CLQ@mail.gmail.com>	<109501d1680d$723c40f0$56b4c2d0$@freeswitch.org>
	<CAMDdSzMh_fsQ+pUNGu3SkrJ3U8BOVvjXnpaJ7GtK5PanmHbH8A@mail.gmail.com>
	<109f01d1680e$d3baf190$7b30d4b0$@freeswitch.org>
Message-ID: <D3A82262-33B7-4F71-9AF6-7BE2E65830E6@insensate.co.uk>

Hi Ken, Conor, folks,
  Along time ago in a land far away ... the [very] early history of SIP was tied up with a bunch of other multimedia SxP things.
The big driver at that time was being able to do conferencing and media distribution -- after all, voice calls could be done with H.323 et al. The SDP (and SIP main part headers) were quite simple and small [**]. This was a message based scheme, and UDP is a messaging transport, as opposed to TCP which is a stream transport.

IIRC, mapping from PSTN schemes (again, message-based systems) to UDP seemed simpler.
TCP required maintaining transport session state in gateways, and the stacks in those gateways were primitive, to say the least.

Despite that, folk pushing for TCP to be mandatory were told that it was considered 2nd class and should not be mandatory to implement); that was in '97 as I recall.

Then (late 98 -> 2001) cable labs & 3GPP decided SIP was easier to bend to their will than H.323/224/..., and the number of headers grew like topsy, the complexity of the maintained state just kept on building, and we ran into fragment problems.
Quick fix was header compression, but that ran into company-political issues in 3GPP and anyway couldn't keep up with the 5,000 new headers there seemed to be per week. THEN there was a drift away from UDP and towards TCP for purely practical reasons, and TCP became mandatory to implement (but NOT, of course mandatory to use, as there were any number of bits of kit out there that didn't have support for it :).

Long story, but in short -- with the continued introduction of bloat (e.g., IMHO all the web RTC driven stuff) UDP is getting VERY tight on MTU limits. That shouldn't be a problem but is because frags are not dealt with well by end systems (as customer router/end system IP stacks tend to be nasty brutish and short on development).

SO ... TCP has advantages (as long as your system can handle many parallel TCP sessions), is marginally slower on initial set up, but doesn't have to maintain the t30 et al timer stuff. From memory, getting the TCP stack tweaked for ultra-high load systems was a pain and led to obscure behaviour, but available TCP stacks seem generally better now.
UDP was simpler to map to message based systems at gateways, didn't have to use good IP stacks as you were rolling your own logic, but given the lard that is SIP/SDP now, that's the least of your coding worries.

For carriers, I understand why they have a reflex against maintaining state, and they're using kit that is "mature". It's hard to justify replacing kit that's familiar, has a management UI your staff know, and had its costs amortised away years ago; VoIP is not a high profit service so the bean counters WILL ask.

=> TCP may be 'better', but UDP is in kit that isn't going away soon.

all the best,
  Lawrence


**: Remember, at the time ('97-'98) Henning Schulzrinne was teaching at Columbia University a post-grad course on IP comms during which he gave "implement a SIP-based voice call system" as a [two week] homework exercise, followed by interops between the clients. It had to be simple (and he was a "hard task master" [or words to that effect]; he knew that UDP forced all the timer logic to be coded as well).

On 15 Feb 2016, at 16:35, Ken Rice <krice at freeswitch.org> wrote:
> The problem still exists for expanding SDPs? using TCP to the user/device then trying to send the same thing out to the carrier over UDP is what was causing the problem in the first place? so the decision was made to prevent those problems we?ll only offer what the device offers and not expand the number of codecs even further increasing the already bloated SDPs to the point where they fragment over UDP and get dropped?
>  
> So is TCP better for some things, yes it is, however, the lack of market wide support for it with carriers makes it a pain in the ass even tho the RFCs specifically say you MUST support both UDP and TCP for SIP, but certain VoIP softwares out there only implemented UDP many years ago and now we?re stuck with that legacy
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Colton Conor
> Sent: Monday, February 15, 2016 10:30 AM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode
>  
> True,
>  
> But freeswitch talking to the carriers is almost always UPD.
>  
> However, freeswitch talking to the clients I would say TCP would be idea. So its almost like freeswitch is trancoding from TCP to UDP too :)
>  
> On Mon, Feb 15, 2016 at 10:25 AM, Ken Rice <krice at freeswitch.org> wrote:
> The problem isn?t necessarily the devices, but there is also the carriers?  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Colton Conor
> Sent: Monday, February 15, 2016 10:04 AM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode
>  
> So if the device supports TCP, is there any reason not to use TCP. AKA is there any reason to keep on using UDP. TCP seems superior. 
>  
> On Mon, Feb 15, 2016 at 9:33 AM, Michael Jerris <mike at jerris.com> wrote:
> any device that even remotely follows sip
> specs supports TCP.  Most phones I have seen do.
> 
> 
> On Sunday, February 14, 2016, Colton Conor <colton.conor at gmail.com> wrote:
> So is TCP the preferred method of doing SIP these days? I like TCP with endpoints as they always break through firewalls and we never seem to have in issue with TCP. However UDP is a headache. So if you have the choice why not do TCP? I realize some devices only support UDP, but the majority of SIP phones out there today do support TCP.
>  
> Plus if you use TLS for encryption and security then you are already using TCP right? 
>  
> On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice <krice at freeswitch.org> wrote:
> This behavior changed a while ago. This was dictates by ever growing SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal with fragmentation and everyone refuses to fully implement sip over tcp for some reason even tho a ton of things support it and the RFCs require it
> 
> Sent from my iPhone
> 
> > On Feb 14, 2016, at 3:37 PM, Rajil Saraswat <rajil.s at gmail.com> wrote:
> >
> > Thanks, after setting media_mix_inbound_outbound_codecs=true,
> > transcoding happens automatically.  I remember not setting this
> > variable in other installations and transcoding used to work out of
> > the box. Is media_mix_inbound_outbound_codecs=true default in
> > Freeswitch?
> >
> >> On 14 February 2016 at 13:56, Russell Treleaven <rtreleaven at bunnykick.ca> wrote:
> >> fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation
> >>
> >>> On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <rajil.s at gmail.com> wrote:
> >>>
> >>> The siptrace is at http://pastebin.com/xiGqtj1Y
> >>>
> >>> The call is being made from 303 (Android/CSipsimple with OPUS codec)
> >>> to 208 (pjsua test client with PCMU codec). The error is on line 545.
> >>>
> >>> On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz at gmail.com>
> >>> wrote:
> >>>> How you originate the call? Is a bridge? >From which phone?
> >>>>
> >>>> Also, please pastebin the complete sip trace (from start of leg A to end
> >>>> of
> >>>> both legs) and put here a link to pastebin
> >>>>
> >>>> Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s at gmail.com> ha scritto:
> >>>>>
> >>>>> Hello,
> >>>>>
> >>>>> I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
> >>>>> When I call to this remote sip phone i get a 406 error that opus is
> >>>>> not supported as shown by the sip trace below. However, if I force the
> >>>>> codec to absolute like this
> >>>>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5
> >>>>> the call works fine.
> >>>>>
> >>>>> Is there anyway I can make FreeSWITCH to automatically transcode
> >>>>> without forcing the codec string in the dial plan?
> >>>>>
> >>>>> The codec preferences is set as  global_codec_prefs=OPUS,PCMU,PCMA and
> >>>>> outbound_codec_prefs=PCMU,PCMA,GSM
> >>>>>
> >>>>> ---------------------------siptrace--------------------------------
> >>>>>
> >>>>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
> >>>>>
> >>>>>
> >>>>> ------------------------------------------------------------------------
> >>>>>   SIP/2.0 406 Not Acceptable
> >>>>>   Via: SIP/2.0/UDP
> >>>>>
> >>>>>
> >>>>> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
> >>>>>   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
> >>>>>   From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
> >>>>>   To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
> >>>>>   CSeq: 87372504 INVITE
> >>>>>   Content-Length:  0
> >>>>>
> >>>>>
> >>>>>
> >>>>> ------------------------------------------------------------------------
> >>>>> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:
> >>>>>
> >>>>>
> >>>>> ------------------------------------------------------------------------
> >>>>>   ACK sip:303 at 192.168.1.5 SIP/2.0
> >>>>>   Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
> >>>>>   Max-Forwards: 68
> >>>>>   From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
> >>>>>   To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
> >>>>>   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
> >>>>>   CSeq: 87372504 ACK
> >>>>>   Content-Length: 0
> >>>>>
> >>>>>
> >>>>>
> >>>>> ------------------------------------------------------------------------
> >>>>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
> >>>>> sofia/internal/303 at 192.168.1.5 entering state [terminated][406]
> >>>>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
> >>>>> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA]
> >>>>> [SERVICE_NOT_IMPLEMENTED]
> >>>>>
> >>>>> Thanks
> >>>>> Rajil
> >>>>>
> >>>>>
> >>>>> _________________________________________________________________________
> >>>>> Professional FreeSWITCH Consulting Services:
> >>>>> consulting at freeswitch.org
> >>>>> http://www.freeswitchsolutions.com
> >>>>>
> >>>>> Official FreeSWITCH Sites
> >>>>> http://www.freeswitch.org
> >>>>> http://confluence.freeswitch.org
> >>>>> http://www.cluecon.com
> >>>>>
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>
> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>
> >>>>
> >>>>
> >>>> _________________________________________________________________________
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>>
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://confluence.freeswitch.org
> >>>> http://www.cluecon.com
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>
> >>> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://confluence.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Mon Feb 15 21:53:12 2016
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 15 Feb 2016 12:53:12 -0600
Subject: [Freeswitch-users] TCP vs UDP (was Re: Freeswitch
	doesnt	transcode)
In-Reply-To: <D3A82262-33B7-4F71-9AF6-7BE2E65830E6@insensate.co.uk>
References: <CAFoaQoAoVLu7ni7x02Y-2hS3nvjf_vdphJkLqt4+mxbhx7=g+w@mail.gmail.com>	<CALXCt0qKR1OusCr0i_2mrOS3i8fzY-sbu4Q_FdCuX1EBSgPA6A@mail.gmail.com>	<CAFoaQoASOwftNQXF5JzMvN76WwvDaf01MTz_C8CR-Y-KKVHzeA@mail.gmail.com>	<CAHzfiZ5ckpCetwT6HYAJiH-GRJ32FgQObqCuO0LPK_wSEff74Q@mail.gmail.com>	<CAFoaQoCkvqRUT4R2LFBFKMLs9YVjp5US_kekr1EKTHUO_Z99oA@mail.gmail.com>	<87D80ECA-87A9-4824-B9E7-4A1EE91AB4DC@freeswitch.org>	<CAMDdSzNiDFhza8pFO=Udmi+0ptgu8uM8rbLXjubo0qFkDiGykw@mail.gmail.com>	<CAEwTsAj7jy7uPzESmZamhNZF76uVFBp8YJ63sU_2jFyH7vayHQ@mail.gmail.com>	<CAMDdSzM3aVHTwi5DAGgfdqFEt8FO60YPfhPDYQUyrsyVYP2CLQ@mail.gmail.com>	<109501d1680d$723c40f0$56b4c2d0$@freeswitch.org>	<CAMDdSzMh_fsQ+pUNGu3SkrJ3U8BOVvjXnpaJ7GtK5PanmHbH8A@mail.gmail.com>	<109f01d1680e$d3baf190$7b30d4b0$@freeswitch.org>
	<D3A82262-33B7-4F71-9AF6-7BE2E65830E6@insensate.co.uk>
Message-ID: <113801d16822$1f4cf910$5de6eb30$@freeswitch.org>

Lawrence,

Well Said! 

There is one upside., atleast Microsoft pushed the TCP stuff hard with Lync
so maybe we'll start seeing more traction there...

In reguards to the WebRTC stuff, imho SIP over WebRTC is a bit heavy
handed... something simple like Verto provides a lower overhead (in the
browser) and allows for push/pull eventing... wish we would see wider
adoption of such things in the near to mid terms

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lawrence
Conroy
Sent: Monday, February 15, 2016 12:45 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] TCP vs UDP (was Re: Freeswitch doesnt
transcode)

Hi Ken, Conor, folks,
  Along time ago in a land far away ... the [very] early history of SIP was
tied up with a bunch of other multimedia SxP things.
The big driver at that time was being able to do conferencing and media
distribution -- after all, voice calls could be done with H.323 et al. The
SDP (and SIP main part headers) were quite simple and small [**]. This was a
message based scheme, and UDP is a messaging transport, as opposed to TCP
which is a stream transport.

IIRC, mapping from PSTN schemes (again, message-based systems) to UDP seemed
simpler.
TCP required maintaining transport session state in gateways, and the stacks
in those gateways were primitive, to say the least.

Despite that, folk pushing for TCP to be mandatory were told that it was
considered 2nd class and should not be mandatory to implement); that was in
'97 as I recall.

Then (late 98 -> 2001) cable labs & 3GPP decided SIP was easier to bend to
their will than H.323/224/..., and the number of headers grew like topsy,
the complexity of the maintained state just kept on building, and we ran
into fragment problems.
Quick fix was header compression, but that ran into company-political issues
in 3GPP and anyway couldn't keep up with the 5,000 new headers there seemed
to be per week. THEN there was a drift away from UDP and towards TCP for
purely practical reasons, and TCP became mandatory to implement (but NOT, of
course mandatory to use, as there were any number of bits of kit out there
that didn't have support for it :).

Long story, but in short -- with the continued introduction of bloat (e.g.,
IMHO all the web RTC driven stuff) UDP is getting VERY tight on MTU limits.
That shouldn't be a problem but is because frags are not dealt with well by
end systems (as customer router/end system IP stacks tend to be nasty
brutish and short on development).

SO ... TCP has advantages (as long as your system can handle many parallel
TCP sessions), is marginally slower on initial set up, but doesn't have to
maintain the t30 et al timer stuff. From memory, getting the TCP stack
tweaked for ultra-high load systems was a pain and led to obscure behaviour,
but available TCP stacks seem generally better now.
UDP was simpler to map to message based systems at gateways, didn't have to
use good IP stacks as you were rolling your own logic, but given the lard
that is SIP/SDP now, that's the least of your coding worries.

For carriers, I understand why they have a reflex against maintaining state,
and they're using kit that is "mature". It's hard to justify replacing kit
that's familiar, has a management UI your staff know, and had its costs
amortised away years ago; VoIP is not a high profit service so the bean
counters WILL ask.

=> TCP may be 'better', but UDP is in kit that isn't going away soon.

all the best,
  Lawrence


**: Remember, at the time ('97-'98) Henning Schulzrinne was teaching at
Columbia University a post-grad course on IP comms during which he gave
"implement a SIP-based voice call system" as a [two week] homework exercise,
followed by interops between the clients. It had to be simple (and he was a
"hard task master" [or words to that effect]; he knew that UDP forced all
the timer logic to be coded as well).

On 15 Feb 2016, at 16:35, Ken Rice <krice at freeswitch.org> wrote:
> The problem still exists for expanding SDPs. using TCP to the 
> user/device then trying to send the same thing out to the carrier over 
> UDP is what was causing the problem in the first place. so the 
> decision was made to prevent those problems we'll only offer what the 
> device offers and not expand the number of codecs even further 
> increasing the already bloated SDPs to the point where they fragment 
> over UDP and get dropped.
>  
> So is TCP better for some things, yes it is, however, the lack of 
> market wide support for it with carriers makes it a pain in the ass 
> even tho the RFCs specifically say you MUST support both UDP and TCP 
> for SIP, but certain VoIP softwares out there only implemented UDP 
> many years ago and now we're stuck with that legacy
>  
> From: freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of 
> Colton Conor
> Sent: Monday, February 15, 2016 10:30 AM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode
>  
> True,
>  
> But freeswitch talking to the carriers is almost always UPD.
>  
> However, freeswitch talking to the clients I would say TCP would be 
> idea. So its almost like freeswitch is trancoding from TCP to UDP too 
> :)
>  
> On Mon, Feb 15, 2016 at 10:25 AM, Ken Rice <krice at freeswitch.org> wrote:
> The problem isn't necessarily the devices, but there is also the 
> carriers.
>  
> From: freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of 
> Colton Conor
> Sent: Monday, February 15, 2016 10:04 AM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode
>  
> So if the device supports TCP, is there any reason not to use TCP. AKA is
there any reason to keep on using UDP. TCP seems superior. 
>  
> On Mon, Feb 15, 2016 at 9:33 AM, Michael Jerris <mike at jerris.com> wrote:
> any device that even remotely follows sip specs supports TCP.  Most 
> phones I have seen do.
> 
> 
> On Sunday, February 14, 2016, Colton Conor <colton.conor at gmail.com> wrote:
> So is TCP the preferred method of doing SIP these days? I like TCP with
endpoints as they always break through firewalls and we never seem to have
in issue with TCP. However UDP is a headache. So if you have the choice why
not do TCP? I realize some devices only support UDP, but the majority of SIP
phones out there today do support TCP.
>  
> Plus if you use TLS for encryption and security then you are already using
TCP right? 
>  
> On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice <krice at freeswitch.org> wrote:
> This behavior changed a while ago. This was dictates by ever growing 
> SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal 
> with fragmentation and everyone refuses to fully implement sip over 
> tcp for some reason even tho a ton of things support it and the RFCs 
> require it
> 
> Sent from my iPhone
> 
> > On Feb 14, 2016, at 3:37 PM, Rajil Saraswat <rajil.s at gmail.com> wrote:
> >
> > Thanks, after setting media_mix_inbound_outbound_codecs=true,
> > transcoding happens automatically.  I remember not setting this 
> > variable in other installations and transcoding used to work out of 
> > the box. Is media_mix_inbound_outbound_codecs=true default in 
> > Freeswitch?
> >
> >> On 14 February 2016 at 13:56, Russell Treleaven
<rtreleaven at bunnykick.ca> wrote:
> >> fyi 
> >> https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiat
> >> ion
> >>
> >>> On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <rajil.s at gmail.com>
wrote:
> >>>
> >>> The siptrace is at http://pastebin.com/xiGqtj1Y
> >>>
> >>> The call is being made from 303 (Android/CSipsimple with OPUS 
> >>> codec) to 208 (pjsua test client with PCMU codec). The error is on
line 545.
> >>>
> >>> On 14 February 2016 at 11:28, Giovanni Maruzzelli 
> >>> <gmaruzz at gmail.com>
> >>> wrote:
> >>>> How you originate the call? Is a bridge? >From which phone?
> >>>>
> >>>> Also, please pastebin the complete sip trace (from start of leg A 
> >>>> to end of both legs) and put here a link to pastebin
> >>>>
> >>>> Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s at gmail.com> ha
scritto:
> >>>>>
> >>>>> Hello,
> >>>>>
> >>>>> I have a remote sip phone (Linksys SPA3102) which only supports
PCMU.
> >>>>> When I call to this remote sip phone i get a 406 error that opus 
> >>>>> is not supported as shown by the sip trace below. However, if I 
> >>>>> force the codec to absolute like this
> >>>>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.
> >>>>> 5
> >>>>> the call works fine.
> >>>>>
> >>>>> Is there anyway I can make FreeSWITCH to automatically transcode 
> >>>>> without forcing the codec string in the dial plan?
> >>>>>
> >>>>> The codec preferences is set as  
> >>>>> global_codec_prefs=OPUS,PCMU,PCMA and 
> >>>>> outbound_codec_prefs=PCMU,PCMA,GSM
> >>>>>
> >>>>> ---------------------------siptrace-----------------------------
> >>>>> ---
> >>>>>
> >>>>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
> >>>>>
> >>>>>
> >>>>>
------------------------------------------------------------------------
> >>>>>   SIP/2.0 406 Not Acceptable
> >>>>>   Via: SIP/2.0/UDP
> >>>>>
> >>>>>
> >>>>>
192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
> >>>>>   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
> >>>>>   From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
> >>>>>   To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
> >>>>>   CSeq: 87372504 INVITE
> >>>>>   Content-Length:  0
> >>>>>
> >>>>>
> >>>>>
> >>>>> ----------------------------------------------------------------
> >>>>> -------- send 324 bytes to udp/[192.168.1.5]:5060 at 
> >>>>> 08:02:16.368591:
> >>>>>
> >>>>>
> >>>>>
------------------------------------------------------------------------
> >>>>>   ACK sip:303 at 192.168.1.5 SIP/2.0
> >>>>>   Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
> >>>>>   Max-Forwards: 68
> >>>>>   From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
> >>>>>   To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
> >>>>>   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
> >>>>>   CSeq: 87372504 ACK
> >>>>>   Content-Length: 0
> >>>>>
> >>>>>
> >>>>>
> >>>>> ----------------------------------------------------------------
> >>>>> --------
> >>>>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
> >>>>> sofia/internal/303 at 192.168.1.5 entering state [terminated][406]
> >>>>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
> >>>>> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA] 
> >>>>> [SERVICE_NOT_IMPLEMENTED]
> >>>>>
> >>>>> Thanks
> >>>>> Rajil
> >>>>>
> >>>>>
> >>>>> ________________________________________________________________
> >>>>> _________ Professional FreeSWITCH Consulting Services:
> >>>>> consulting at freeswitch.org
> >>>>> http://www.freeswitchsolutions.com
> >>>>>
> >>>>> Official FreeSWITCH Sites
> >>>>> http://www.freeswitch.org
> >>>>> http://confluence.freeswitch.org http://www.cluecon.com
> >>>>>
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>
> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswit
> >>>>> ch-users
> >>>>> http://www.freeswitch.org
> >>>>
> >>>>
> >>>>
> >>>> _________________________________________________________________
> >>>> ________ Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>>
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://confluence.freeswitch.org
> >>>> http://www.cluecon.com
> >>>>
> >>>> FreeSWITCH-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
> >>>
> >>> __________________________________________________________________
> >>> _______ Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://confluence.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch
> >>> -users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >> ___________________________________________________________________
> >> ______ Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >
> > ____________________________________________________________________
> > _____ Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u
> > sers
> > http://www.freeswitch.org
> 
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>  
> 
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>  
> 
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>  
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org


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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at 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  Mon Feb 15 22:26:30 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Feb 2016 13:26:30 -0600
Subject: [Freeswitch-users] TCP vs UDP (was Re: Freeswitch doesnt
	transcode)
In-Reply-To: <113801d16822$1f4cf910$5de6eb30$@freeswitch.org>
References: <CAFoaQoAoVLu7ni7x02Y-2hS3nvjf_vdphJkLqt4+mxbhx7=g+w@mail.gmail.com>
	<CALXCt0qKR1OusCr0i_2mrOS3i8fzY-sbu4Q_FdCuX1EBSgPA6A@mail.gmail.com>
	<CAFoaQoASOwftNQXF5JzMvN76WwvDaf01MTz_C8CR-Y-KKVHzeA@mail.gmail.com>
	<CAHzfiZ5ckpCetwT6HYAJiH-GRJ32FgQObqCuO0LPK_wSEff74Q@mail.gmail.com>
	<CAFoaQoCkvqRUT4R2LFBFKMLs9YVjp5US_kekr1EKTHUO_Z99oA@mail.gmail.com>
	<87D80ECA-87A9-4824-B9E7-4A1EE91AB4DC@freeswitch.org>
	<CAMDdSzNiDFhza8pFO=Udmi+0ptgu8uM8rbLXjubo0qFkDiGykw@mail.gmail.com>
	<CAEwTsAj7jy7uPzESmZamhNZF76uVFBp8YJ63sU_2jFyH7vayHQ@mail.gmail.com>
	<CAMDdSzM3aVHTwi5DAGgfdqFEt8FO60YPfhPDYQUyrsyVYP2CLQ@mail.gmail.com>
	<109501d1680d$723c40f0$56b4c2d0$@freeswitch.org>
	<CAMDdSzMh_fsQ+pUNGu3SkrJ3U8BOVvjXnpaJ7GtK5PanmHbH8A@mail.gmail.com>
	<109f01d1680e$d3baf190$7b30d4b0$@freeswitch.org>
	<D3A82262-33B7-4F71-9AF6-7BE2E65830E6@insensate.co.uk>
	<113801d16822$1f4cf910$5de6eb30$@freeswitch.org>
Message-ID: <CAKbxfG-LMDUT1vpssc4mgctMFm3avtSUS6H7i0TXZLB7bB77gQ@mail.gmail.com>

With SDP getting bigger and bigger with presence packates and large invites
with many codecs or video and WebRTC TCP will become mandatory.
The spec on when to use TCP is very arcane.

Use UDP first unless the packet is > MTU, change to TCP.  If the TCP times
out (1 to 10 min) retry UDP anyway.
With some fun mixed in like you MUST be under the MTU and you also MUST
support packets over udp up to 64kb.

Implementing that used to cause communications with asterisk to take
forever because they only did UDP so bigger SDP packets would timeout on
TCP first and everyone called it a bug.

If you anticipate using presence or really big packets use TCP.  If you use
WebRTC its already TCP.




On Mon, Feb 15, 2016 at 12:53 PM, Ken Rice <krice at freeswitch.org> wrote:

> Lawrence,
>
> Well Said!
>
> There is one upside., atleast Microsoft pushed the TCP stuff hard with Lync
> so maybe we'll start seeing more traction there...
>
> In reguards to the WebRTC stuff, imho SIP over WebRTC is a bit heavy
> handed... something simple like Verto provides a lower overhead (in the
> browser) and allows for push/pull eventing... wish we would see wider
> adoption of such things in the near to mid terms
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Lawrence
> Conroy
> Sent: Monday, February 15, 2016 12:45 PM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] TCP vs UDP (was Re: Freeswitch doesnt
> transcode)
>
> Hi Ken, Conor, folks,
>   Along time ago in a land far away ... the [very] early history of SIP was
> tied up with a bunch of other multimedia SxP things.
> The big driver at that time was being able to do conferencing and media
> distribution -- after all, voice calls could be done with H.323 et al. The
> SDP (and SIP main part headers) were quite simple and small [**]. This was
> a
> message based scheme, and UDP is a messaging transport, as opposed to TCP
> which is a stream transport.
>
> IIRC, mapping from PSTN schemes (again, message-based systems) to UDP
> seemed
> simpler.
> TCP required maintaining transport session state in gateways, and the
> stacks
> in those gateways were primitive, to say the least.
>
> Despite that, folk pushing for TCP to be mandatory were told that it was
> considered 2nd class and should not be mandatory to implement); that was in
> '97 as I recall.
>
> Then (late 98 -> 2001) cable labs & 3GPP decided SIP was easier to bend to
> their will than H.323/224/..., and the number of headers grew like topsy,
> the complexity of the maintained state just kept on building, and we ran
> into fragment problems.
> Quick fix was header compression, but that ran into company-political
> issues
> in 3GPP and anyway couldn't keep up with the 5,000 new headers there seemed
> to be per week. THEN there was a drift away from UDP and towards TCP for
> purely practical reasons, and TCP became mandatory to implement (but NOT,
> of
> course mandatory to use, as there were any number of bits of kit out there
> that didn't have support for it :).
>
> Long story, but in short -- with the continued introduction of bloat (e.g.,
> IMHO all the web RTC driven stuff) UDP is getting VERY tight on MTU limits.
> That shouldn't be a problem but is because frags are not dealt with well by
> end systems (as customer router/end system IP stacks tend to be nasty
> brutish and short on development).
>
> SO ... TCP has advantages (as long as your system can handle many parallel
> TCP sessions), is marginally slower on initial set up, but doesn't have to
> maintain the t30 et al timer stuff. From memory, getting the TCP stack
> tweaked for ultra-high load systems was a pain and led to obscure
> behaviour,
> but available TCP stacks seem generally better now.
> UDP was simpler to map to message based systems at gateways, didn't have to
> use good IP stacks as you were rolling your own logic, but given the lard
> that is SIP/SDP now, that's the least of your coding worries.
>
> For carriers, I understand why they have a reflex against maintaining
> state,
> and they're using kit that is "mature". It's hard to justify replacing kit
> that's familiar, has a management UI your staff know, and had its costs
> amortised away years ago; VoIP is not a high profit service so the bean
> counters WILL ask.
>
> => TCP may be 'better', but UDP is in kit that isn't going away soon.
>
> all the best,
>   Lawrence
>
>
> **: Remember, at the time ('97-'98) Henning Schulzrinne was teaching at
> Columbia University a post-grad course on IP comms during which he gave
> "implement a SIP-based voice call system" as a [two week] homework
> exercise,
> followed by interops between the clients. It had to be simple (and he was a
> "hard task master" [or words to that effect]; he knew that UDP forced all
> the timer logic to be coded as well).
>
> On 15 Feb 2016, at 16:35, Ken Rice <krice at freeswitch.org> wrote:
> > The problem still exists for expanding SDPs. using TCP to the
> > user/device then trying to send the same thing out to the carrier over
> > UDP is what was causing the problem in the first place. so the
> > decision was made to prevent those problems we'll only offer what the
> > device offers and not expand the number of codecs even further
> > increasing the already bloated SDPs to the point where they fragment
> > over UDP and get dropped.
> >
> > So is TCP better for some things, yes it is, however, the lack of
> > market wide support for it with carriers makes it a pain in the ass
> > even tho the RFCs specifically say you MUST support both UDP and TCP
> > for SIP, but certain VoIP softwares out there only implemented UDP
> > many years ago and now we're stuck with that legacy
> >
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> > Colton Conor
> > Sent: Monday, February 15, 2016 10:30 AM
> > To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> > Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode
> >
> > True,
> >
> > But freeswitch talking to the carriers is almost always UPD.
> >
> > However, freeswitch talking to the clients I would say TCP would be
> > idea. So its almost like freeswitch is trancoding from TCP to UDP too
> > :)
> >
> > On Mon, Feb 15, 2016 at 10:25 AM, Ken Rice <krice at freeswitch.org> wrote:
> > The problem isn't necessarily the devices, but there is also the
> > carriers.
> >
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> > Colton Conor
> > Sent: Monday, February 15, 2016 10:04 AM
> > To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> > Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode
> >
> > So if the device supports TCP, is there any reason not to use TCP. AKA is
> there any reason to keep on using UDP. TCP seems superior.
> >
> > On Mon, Feb 15, 2016 at 9:33 AM, Michael Jerris <mike at jerris.com> wrote:
> > any device that even remotely follows sip specs supports TCP.  Most
> > phones I have seen do.
> >
> >
> > On Sunday, February 14, 2016, Colton Conor <colton.conor at gmail.com>
> wrote:
> > So is TCP the preferred method of doing SIP these days? I like TCP with
> endpoints as they always break through firewalls and we never seem to have
> in issue with TCP. However UDP is a headache. So if you have the choice why
> not do TCP? I realize some devices only support UDP, but the majority of
> SIP
> phones out there today do support TCP.
> >
> > Plus if you use TLS for encryption and security then you are already
> using
> TCP right?
> >
> > On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice <krice at freeswitch.org> wrote:
> > This behavior changed a while ago. This was dictates by ever growing
> > SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal
> > with fragmentation and everyone refuses to fully implement sip over
> > tcp for some reason even tho a ton of things support it and the RFCs
> > require it
> >
> > Sent from my iPhone
> >
> > > On Feb 14, 2016, at 3:37 PM, Rajil Saraswat <rajil.s at gmail.com> wrote:
> > >
> > > Thanks, after setting media_mix_inbound_outbound_codecs=true,
> > > transcoding happens automatically.  I remember not setting this
> > > variable in other installations and transcoding used to work out of
> > > the box. Is media_mix_inbound_outbound_codecs=true default in
> > > Freeswitch?
> > >
> > >> On 14 February 2016 at 13:56, Russell Treleaven
> <rtreleaven at bunnykick.ca> wrote:
> > >> fyi
> > >> https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiat
> > >> ion
> > >>
> > >>> On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <rajil.s at gmail.com>
> wrote:
> > >>>
> > >>> The siptrace is at http://pastebin.com/xiGqtj1Y
> > >>>
> > >>> The call is being made from 303 (Android/CSipsimple with OPUS
> > >>> codec) to 208 (pjsua test client with PCMU codec). The error is on
> line 545.
> > >>>
> > >>> On 14 February 2016 at 11:28, Giovanni Maruzzelli
> > >>> <gmaruzz at gmail.com>
> > >>> wrote:
> > >>>> How you originate the call? Is a bridge? >From which phone?
> > >>>>
> > >>>> Also, please pastebin the complete sip trace (from start of leg A
> > >>>> to end of both legs) and put here a link to pastebin
> > >>>>
> > >>>> Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s at gmail.com> ha
> scritto:
> > >>>>>
> > >>>>> Hello,
> > >>>>>
> > >>>>> I have a remote sip phone (Linksys SPA3102) which only supports
> PCMU.
> > >>>>> When I call to this remote sip phone i get a 406 error that opus
> > >>>>> is not supported as shown by the sip trace below. However, if I
> > >>>>> force the codec to absolute like this
> > >>>>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.
> > >>>>> 5
> > >>>>> the call works fine.
> > >>>>>
> > >>>>> Is there anyway I can make FreeSWITCH to automatically transcode
> > >>>>> without forcing the codec string in the dial plan?
> > >>>>>
> > >>>>> The codec preferences is set as
> > >>>>> global_codec_prefs=OPUS,PCMU,PCMA and
> > >>>>> outbound_codec_prefs=PCMU,PCMA,GSM
> > >>>>>
> > >>>>> ---------------------------siptrace-----------------------------
> > >>>>> ---
> > >>>>>
> > >>>>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
> > >>>>>
> > >>>>>
> > >>>>>
> ------------------------------------------------------------------------
> > >>>>>   SIP/2.0 406 Not Acceptable
> > >>>>>   Via: SIP/2.0/UDP
> > >>>>>
> > >>>>>
> > >>>>>
> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
> > >>>>>   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
> > >>>>>   From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
> > >>>>>   To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
> > >>>>>   CSeq: 87372504 INVITE
> > >>>>>   Content-Length:  0
> > >>>>>
> > >>>>>
> > >>>>>
> > >>>>> ----------------------------------------------------------------
> > >>>>> -------- send 324 bytes to udp/[192.168.1.5]:5060 at
> > >>>>> 08:02:16.368591:
> > >>>>>
> > >>>>>
> > >>>>>
> ------------------------------------------------------------------------
> > >>>>>   ACK sip:303 at 192.168.1.5 SIP/2.0
> > >>>>>   Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
> > >>>>>   Max-Forwards: 68
> > >>>>>   From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
> > >>>>>   To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
> > >>>>>   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
> > >>>>>   CSeq: 87372504 ACK
> > >>>>>   Content-Length: 0
> > >>>>>
> > >>>>>
> > >>>>>
> > >>>>> ----------------------------------------------------------------
> > >>>>> --------
> > >>>>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
> > >>>>> sofia/internal/303 at 192.168.1.5 entering state [terminated][406]
> > >>>>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
> > >>>>> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA]
> > >>>>> [SERVICE_NOT_IMPLEMENTED]
> > >>>>>
> > >>>>> Thanks
> > >>>>> Rajil
> > >>>>>
> > >>>>>
> > >>>>> ________________________________________________________________
> > >>>>> _________ Professional FreeSWITCH Consulting Services:
> > >>>>> consulting at freeswitch.org
> > >>>>> http://www.freeswitchsolutions.com
> > >>>>>
> > >>>>> Official FreeSWITCH Sites
> > >>>>> http://www.freeswitch.org
> > >>>>> http://confluence.freeswitch.org http://www.cluecon.com
> > >>>>>
> > >>>>> FreeSWITCH-users mailing list
> > >>>>> FreeSWITCH-users at lists.freeswitch.org
> > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >>>>>
> > >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswit
> > >>>>> ch-users
> > >>>>> http://www.freeswitch.org
> > >>>>
> > >>>>
> > >>>>
> > >>>> _________________________________________________________________
> > >>>> ________ Professional FreeSWITCH Consulting Services:
> > >>>> consulting at freeswitch.org
> > >>>> http://www.freeswitchsolutions.com
> > >>>>
> > >>>> Official FreeSWITCH Sites
> > >>>> http://www.freeswitch.org
> > >>>> http://confluence.freeswitch.org
> > >>>> http://www.cluecon.com
> > >>>>
> > >>>> FreeSWITCH-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
> > >>>
> > >>> __________________________________________________________________
> > >>> _______ Professional FreeSWITCH Consulting Services:
> > >>> consulting at freeswitch.org
> > >>> http://www.freeswitchsolutions.com
> > >>>
> > >>> Official FreeSWITCH Sites
> > >>> http://www.freeswitch.org
> > >>> http://confluence.freeswitch.org
> > >>> http://www.cluecon.com
> > >>>
> > >>> FreeSWITCH-users mailing list
> > >>> FreeSWITCH-users at lists.freeswitch.org
> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch
> > >>> -users
> > >>> http://www.freeswitch.org
> > >>
> > >>
> > >>
> > >> ___________________________________________________________________
> > >> ______ Professional FreeSWITCH Consulting Services:
> > >> consulting at freeswitch.org
> > >> http://www.freeswitchsolutions.com
> > >>
> > >> Official FreeSWITCH Sites
> > >> http://www.freeswitch.org
> > >> http://confluence.freeswitch.org
> > >> http://www.cluecon.com
> > >>
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > >> users
> > >> http://www.freeswitch.org
> > >
> > > ____________________________________________________________________
> > > _____ Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://confluence.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u
> > > sers
> > > http://www.freeswitch.org
> >
> > ______________________________________________________________________
> > ___ Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> > rs
> > http://www.freeswitch.org
> >
> >
> > ______________________________________________________________________
> > ___ Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> > rs
> > http://www.freeswitch.org
> >
> >
> > ______________________________________________________________________
> > ___ Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> > rs
> > http://www.freeswitch.org
> >
> > ______________________________________________________________________
> > ___ Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> > rs
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 brian at freeswitch.org  Mon Feb 15 22:35:11 2016
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Feb 2016 13:35:11 -0600
Subject: [Freeswitch-users] Freeswitch doesnt transcode
In-Reply-To: <CAAemr3y16uriPOdVcErFvQtupMawK67-Nc_BCNsJ3o4LK1x9FA@mail.gmail.com>
References: <CAFoaQoAoVLu7ni7x02Y-2hS3nvjf_vdphJkLqt4+mxbhx7=g+w@mail.gmail.com>
	<CALXCt0qKR1OusCr0i_2mrOS3i8fzY-sbu4Q_FdCuX1EBSgPA6A@mail.gmail.com>
	<CAFoaQoASOwftNQXF5JzMvN76WwvDaf01MTz_C8CR-Y-KKVHzeA@mail.gmail.com>
	<CAHzfiZ5ckpCetwT6HYAJiH-GRJ32FgQObqCuO0LPK_wSEff74Q@mail.gmail.com>
	<CAAemr3y16uriPOdVcErFvQtupMawK67-Nc_BCNsJ3o4LK1x9FA@mail.gmail.com>
Message-ID: <CAEJMVkDs3ci4U2HopNYQnEjqkJ4ZB9Mp7nVRsqxxHOOfnthMjg@mail.gmail.com>

google freeswitch behavior change

we did send out warnings, change logged it, WIR'ed the change too.  ;)

Thanks,

On Mon, Feb 15, 2016 at 12:27 PM, Bob Hartwig <bobjectsfreeswitch at gmail.com>
wrote:

> On Sun, Feb 14, 2016 at 1:56 PM, Russell Treleaven <
> rtreleaven at bunnykick.ca> wrote:
>
>> fyi
>> https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation
>>
>>
> That is a really lucid and complete description of CODEC negotiation in
> FS.  Thanks for the link, and thanks to the original author for writing it.
>
> Bob
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

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 krice at freeswitch.org  Mon Feb 15 22:48:31 2016
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 15 Feb 2016 13:48:31 -0600
Subject: [Freeswitch-users] Freeswitch doesnt transcode
In-Reply-To: <CAEJMVkDs3ci4U2HopNYQnEjqkJ4ZB9Mp7nVRsqxxHOOfnthMjg@mail.gmail.com>
References: <CAFoaQoAoVLu7ni7x02Y-2hS3nvjf_vdphJkLqt4+mxbhx7=g+w@mail.gmail.com>	<CALXCt0qKR1OusCr0i_2mrOS3i8fzY-sbu4Q_FdCuX1EBSgPA6A@mail.gmail.com>	<CAFoaQoASOwftNQXF5JzMvN76WwvDaf01MTz_C8CR-Y-KKVHzeA@mail.gmail.com>	<CAHzfiZ5ckpCetwT6HYAJiH-GRJ32FgQObqCuO0LPK_wSEff74Q@mail.gmail.com>	<CAAemr3y16uriPOdVcErFvQtupMawK67-Nc_BCNsJ3o4LK1x9FA@mail.gmail.com>
	<CAEJMVkDs3ci4U2HopNYQnEjqkJ4ZB9Mp7nVRsqxxHOOfnthMjg@mail.gmail.com>
Message-ID: <118901d16829$d9803160$8c809420$@freeswitch.org>

What? People are supposed to pay attention to announcements that the FreeSWITCH Dev Team makes?!@ what is this tom foolery?

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Monday, February 15, 2016 1:35 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode

 

google freeswitch behavior change

 

we did send out warnings, change logged it, WIR'ed the change too.  ;)

 

Thanks,

 

On Mon, Feb 15, 2016 at 12:27 PM, Bob Hartwig <bobjectsfreeswitch at gmail.com <mailto:bobjectsfreeswitch at gmail.com> > wrote:

On Sun, Feb 14, 2016 at 1:56 PM, Russell Treleaven <rtreleaven at bunnykick.ca <mailto:rtreleaven at bunnykick.ca> > wrote:

fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation

 

 

That is a really lucid and complete description of CODEC negotiation in FS.  Thanks for the link, and thanks to the original author for writing it.

 

Bob

 

 


_________________________________________________________________________
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





 

-- 

Brian West
brian at freeswitch.org <mailto:brian at freeswitch.org> 

  <http://billing.freeswitch.org/templates/default/img/whmcslogo.png> 

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

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 italo at freeswitch.org  Mon Feb 15 22:57:29 2016
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Mon, 15 Feb 2016 16:57:29 -0300
Subject: [Freeswitch-users] Problem with mod_spy
In-Reply-To: <1597675782.6819.1455527545027.JavaMail.root@mailserver.edistar.com>
References: <CABMX_w-SYNKF4_E5Ui+ou0y-qvM8awG0nodCvTgKqRZRtbJ=cw@mail.gmail.com>
	<1597675782.6819.1455527545027.JavaMail.root@mailserver.edistar.com>
Message-ID: <CABMX_w-yfbfctKd2bF=p9E-VB0yi9vV952Z8-dbwcg22FEou2g@mail.gmail.com>

It's probably due to the way you're dialing to your user, I can see in your
log this bridge command: bridge(sofia/internal/4750%myserver)

Just try to dial bridge user/4750 it'll work if 4750 is registered with FS.

On Mon, Feb 15, 2016 at 6:12 AM, Stefano Favaro <stefano.favaro at edistar.com>
wrote:

> Thank you Italo.
>
> This is my log
>
> https://pastebin.freeswitch.org/24555
>
> and this is the id of my user spy call
>
> 94bae5a0-dd2c-4d0e-99fd-8ef81e753227
>
> Stefano
>
> ------------------------------
> *Da: *"?talo Rossi" <italo at freeswitch.org>
> *A: *"FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> *Inviato: *Gioved?, 11 febbraio 2016 18:52:46
> *Oggetto: *Re: [Freeswitch-users] Problem with mod_spy
>
>
> Stefano,
>
> Can you post your debug logs (/log 7)?
>
> Use https://pastebin.freeswitch.org/
>
> On Thu, Feb 11, 2016 at 1:33 PM, Stefano Favaro <
> stefano.favaro at edistar.com> wrote:
>
>> Hello,
>>
>>  I have a problem with the mod_spy module.
>> It seems that it just plays music and do not actually spy.
>> I want to dial an extension with the number of the user I want to spy and wait for a call in or out for that user.
>> Mod_spy application plays music but when a call is handled by the user I can't hear it continues to play the music on hold audio file.
>>
>> This is the dialplan:
>>
>> <extension name="eavesdrop">
>>   <condition field="destination_number" expression="^88(.*)$|^\*0(.*)$">
>>     <action application="answer"/>
>>     <action application="userspy" data="$1 at myserver"/>
>>   </condition>
>> </extension>
>>
>>
>> If I dial 881000, for example, It means I want to spy on user 1000.
>> I have in and out calls from user 1000 but I can't hear.
>> userspy_show in fs_cli, I get :
>> 1000 at myserver : e9165eaf-e8d7-40ad-a8cd-d2963f363684
>>
>> 1 total spy
>>
>> I have FreeSwitch version FreeSWITCH Version 1.4.26-37~64bit (-37 64bit)
>>
>> SF.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From antoine.durant29 at yahoo.fr  Mon Feb 15 23:43:34 2016
From: antoine.durant29 at yahoo.fr (Antoine Durant)
Date: Mon, 15 Feb 2016 20:43:34 +0000 (UTC)
Subject: [Freeswitch-users] multiple internal ip
References: <857665560.7491164.1455569014252.JavaMail.yahoo.ref@mail.yahoo.com>
Message-ID: <857665560.7491164.1455569014252.JavaMail.yahoo@mail.yahoo.com>

 Hi,
I would like to have more internal IP addresses to do some "Multiple Companies".

The WAN IP is used (80.X.X.125) to mount all trunk (1 trunk by company) to the same provider sip.

Is it possible to have an internal IP for each company:
company-a.org => 192.168.1.1
company-b.org => 192.168.1.2
company-c.org => 192.168.1.3
company-X.org => 192.168.1.X

How to use FreeSwitch this way?
Thanks
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From brian at freeswitch.org  Mon Feb 15 23:47:45 2016
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Feb 2016 14:47:45 -0600
Subject: [Freeswitch-users] multiple internal ip
In-Reply-To: <857665560.7491164.1455569014252.JavaMail.yahoo@mail.yahoo.com>
References: <857665560.7491164.1455569014252.JavaMail.yahoo.ref@mail.yahoo.com>
	<857665560.7491164.1455569014252.JavaMail.yahoo@mail.yahoo.com>
Message-ID: <CAEJMVkBhkfaZYXHxUpqKPwE1FjOrcKQ6FwbLV2hyHWdKBQNjFg@mail.gmail.com>

Why not use the DNS name?  Only one IP is needed then.

On Mon, Feb 15, 2016 at 2:43 PM, Antoine Durant <antoine.durant29 at yahoo.fr>
wrote:

> Hi,
>
> I would like to have more internal IP addresses to do some "Multiple
> Companies".
>
> The WAN IP is used (80.X.X.125) to mount all trunk (1 trunk by company) to
> the same provider sip.
>
> Is it possible to have an internal IP for each company:
> company-a.org => 192.168.1.1
> company-b.org => 192.168.1.2
> company-c.org => 192.168.1.3
> company-X.org => 192.168.1.X
>
> How to use FreeSwitch this way?
>
> Thanks
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


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

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 mgg at giagnocavo.net  Tue Feb 16 00:55:53 2016
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Mon, 15 Feb 2016 21:55:53 +0000
Subject: [Freeswitch-users] TCP vs UDP (was Re: Freeswitch
	doesnt	transcode)
In-Reply-To: <CAKbxfG-LMDUT1vpssc4mgctMFm3avtSUS6H7i0TXZLB7bB77gQ@mail.gmail.com>
References: <CAFoaQoAoVLu7ni7x02Y-2hS3nvjf_vdphJkLqt4+mxbhx7=g+w@mail.gmail.com>
	<CALXCt0qKR1OusCr0i_2mrOS3i8fzY-sbu4Q_FdCuX1EBSgPA6A@mail.gmail.com>
	<CAFoaQoASOwftNQXF5JzMvN76WwvDaf01MTz_C8CR-Y-KKVHzeA@mail.gmail.com>
	<CAHzfiZ5ckpCetwT6HYAJiH-GRJ32FgQObqCuO0LPK_wSEff74Q@mail.gmail.com>
	<CAFoaQoCkvqRUT4R2LFBFKMLs9YVjp5US_kekr1EKTHUO_Z99oA@mail.gmail.com>
	<87D80ECA-87A9-4824-B9E7-4A1EE91AB4DC@freeswitch.org>
	<CAMDdSzNiDFhza8pFO=Udmi+0ptgu8uM8rbLXjubo0qFkDiGykw@mail.gmail.com>
	<CAEwTsAj7jy7uPzESmZamhNZF76uVFBp8YJ63sU_2jFyH7vayHQ@mail.gmail.com>
	<CAMDdSzM3aVHTwi5DAGgfdqFEt8FO60YPfhPDYQUyrsyVYP2CLQ@mail.gmail.com>
	<109501d1680d$723c40f0$56b4c2d0$@freeswitch.org>
	<CAMDdSzMh_fsQ+pUNGu3SkrJ3U8BOVvjXnpaJ7GtK5PanmHbH8A@mail.gmail.com>
	<109f01d1680e$d3baf190$7b30d4b0$@freeswitch.org>
	<D3A82262-33B7-4F71-9AF6-7BE2E65830E6@insensate.co.uk>
	<113801d16822$1f4cf910$5de6eb30$@freeswitch.org>
	<CAKbxfG-LMDUT1vpssc4mgctMFm3avtSUS6H7i0TXZLB7bB77gQ@mail.gmail.com>
Message-ID: <DM2PR07MB70209720CDEB8E2FF5E1D9CD4AC0@DM2PR07MB702.namprd07.prod.outlook.com>

And that?s the reason as to why Lync is TCP-only. The commonly large payloads would always be above the SIP MTU meaning they?d need to switch to TCP. Instead of dealing with the pain of doing both (and the joy of adding that to interop tests), it was easier to simply do TCP-only. And really, people should just be using TLS anyways (but UDP+IPSec seems more common).

Also FWIW, it?s easier to defend some attacks on TCP than UDP; at least it eliminates IP spoofing. (Enable syn cookies versus running a most-likely-buggy SIP parser on each message.)

IP frag may not actually be an issue in practise. I looked at the signaling for a VoIP company with tens of thousands of end users. About 1% of all UDP SIP signaling was IP frag?d, though usually around 576 bytes. This had no noticeable impact on service (i.e., I didn?t see any related retransmits). But this is really just an anecdote.

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
Sent: Monday, 15 February, 2016 13:27
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] TCP vs UDP (was Re: Freeswitch doesnt transcode)

With SDP getting bigger and bigger with presence packates and large invites with many codecs or video and WebRTC TCP will become mandatory.
The spec on when to use TCP is very arcane.

Use UDP first unless the packet is > MTU, change to TCP.  If the TCP times out (1 to 10 min) retry UDP anyway.
With some fun mixed in like you MUST be under the MTU and you also MUST support packets over udp up to 64kb.

Implementing that used to cause communications with asterisk to take forever because they only did UDP so bigger SDP packets would timeout on TCP first and everyone called it a bug.

If you anticipate using presence or really big packets use TCP.  If you use WebRTC its already TCP.




On Mon, Feb 15, 2016 at 12:53 PM, Ken Rice <krice at freeswitch.org<mailto:krice at freeswitch.org>> wrote:
Lawrence,

Well Said!

There is one upside., atleast Microsoft pushed the TCP stuff hard with Lync
so maybe we'll start seeing more traction there...

In reguards to the WebRTC stuff, imho SIP over WebRTC is a bit heavy
handed... something simple like Verto provides a lower overhead (in the
browser) and allows for push/pull eventing... wish we would see wider
adoption of such things in the near to mid terms

-----Original Message-----
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 Lawrence
Conroy
Sent: Monday, February 15, 2016 12:45 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] TCP vs UDP (was Re: Freeswitch doesnt
transcode)

Hi Ken, Conor, folks,
  Along time ago in a land far away ... the [very] early history of SIP was
tied up with a bunch of other multimedia SxP things.
The big driver at that time was being able to do conferencing and media
distribution -- after all, voice calls could be done with H.323 et al. The
SDP (and SIP main part headers) were quite simple and small [**]. This was a
message based scheme, and UDP is a messaging transport, as opposed to TCP
which is a stream transport.

IIRC, mapping from PSTN schemes (again, message-based systems) to UDP seemed
simpler.
TCP required maintaining transport session state in gateways, and the stacks
in those gateways were primitive, to say the least.

Despite that, folk pushing for TCP to be mandatory were told that it was
considered 2nd class and should not be mandatory to implement); that was in
'97 as I recall.

Then (late 98 -> 2001) cable labs & 3GPP decided SIP was easier to bend to
their will than H.323/224/..., and the number of headers grew like topsy,
the complexity of the maintained state just kept on building, and we ran
into fragment problems.
Quick fix was header compression, but that ran into company-political issues
in 3GPP and anyway couldn't keep up with the 5,000 new headers there seemed
to be per week. THEN there was a drift away from UDP and towards TCP for
purely practical reasons, and TCP became mandatory to implement (but NOT, of
course mandatory to use, as there were any number of bits of kit out there
that didn't have support for it :).

Long story, but in short -- with the continued introduction of bloat (e.g.,
IMHO all the web RTC driven stuff) UDP is getting VERY tight on MTU limits.
That shouldn't be a problem but is because frags are not dealt with well by
end systems (as customer router/end system IP stacks tend to be nasty
brutish and short on development).

SO ... TCP has advantages (as long as your system can handle many parallel
TCP sessions), is marginally slower on initial set up, but doesn't have to
maintain the t30 et al timer stuff. From memory, getting the TCP stack
tweaked for ultra-high load systems was a pain and led to obscure behaviour,
but available TCP stacks seem generally better now.
UDP was simpler to map to message based systems at gateways, didn't have to
use good IP stacks as you were rolling your own logic, but given the lard
that is SIP/SDP now, that's the least of your coding worries.

For carriers, I understand why they have a reflex against maintaining state,
and they're using kit that is "mature". It's hard to justify replacing kit
that's familiar, has a management UI your staff know, and had its costs
amortised away years ago; VoIP is not a high profit service so the bean
counters WILL ask.

=> TCP may be 'better', but UDP is in kit that isn't going away soon.

all the best,
  Lawrence


**: Remember, at the time ('97-'98) Henning Schulzrinne was teaching at
Columbia University a post-grad course on IP comms during which he gave
"implement a SIP-based voice call system" as a [two week] homework exercise,
followed by interops between the clients. It had to be simple (and he was a
"hard task master" [or words to that effect]; he knew that UDP forced all
the timer logic to be coded as well).

On 15 Feb 2016, at 16:35, Ken Rice <krice at freeswitch.org<mailto:krice at freeswitch.org>> wrote:
> The problem still exists for expanding SDPs. using TCP to the
> user/device then trying to send the same thing out to the carrier over
> UDP is what was causing the problem in the first place. so the
> decision was made to prevent those problems we'll only offer what the
> device offers and not expand the number of codecs even further
> increasing the already bloated SDPs to the point where they fragment
> over UDP and get dropped.
>
> So is TCP better for some things, yes it is, however, the lack of
> market wide support for it with carriers makes it a pain in the ass
> even tho the RFCs specifically say you MUST support both UDP and TCP
> for SIP, but certain VoIP softwares out there only implemented UDP
> many years ago and now we're stuck with that legacy
>
> 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
> Colton Conor
> Sent: Monday, February 15, 2016 10:30 AM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
> Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode
>
> True,
>
> But freeswitch talking to the carriers is almost always UPD.
>
> However, freeswitch talking to the clients I would say TCP would be
> idea. So its almost like freeswitch is trancoding from TCP to UDP too
> :)
>
> On Mon, Feb 15, 2016 at 10:25 AM, Ken Rice <krice at freeswitch.org<mailto:krice at freeswitch.org>> wrote:
> The problem isn't necessarily the devices, but there is also the
> carriers.
>
> 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
> Colton Conor
> Sent: Monday, February 15, 2016 10:04 AM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
> Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode
>
> So if the device supports TCP, is there any reason not to use TCP. AKA is
there any reason to keep on using UDP. TCP seems superior.
>
> On Mon, Feb 15, 2016 at 9:33 AM, Michael Jerris <mike at jerris.com<mailto:mike at jerris.com>> wrote:
> any device that even remotely follows sip specs supports TCP.  Most
> phones I have seen do.
>
>
> On Sunday, February 14, 2016, Colton Conor <colton.conor at gmail.com<mailto:colton.conor at gmail.com>> wrote:
> So is TCP the preferred method of doing SIP these days? I like TCP with
endpoints as they always break through firewalls and we never seem to have
in issue with TCP. However UDP is a headache. So if you have the choice why
not do TCP? I realize some devices only support UDP, but the majority of SIP
phones out there today do support TCP.
>
> Plus if you use TLS for encryption and security then you are already using
TCP right?
>
> On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice <krice at freeswitch.org<mailto:krice at freeswitch.org>> wrote:
> This behavior changed a while ago. This was dictates by ever growing
> SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal
> with fragmentation and everyone refuses to fully implement sip over
> tcp for some reason even tho a ton of things support it and the RFCs
> require it
>
> Sent from my iPhone
>
> > On Feb 14, 2016, at 3:37 PM, Rajil Saraswat <rajil.s at gmail.com<mailto:rajil.s at gmail.com>> wrote:
> >
> > Thanks, after setting media_mix_inbound_outbound_codecs=true,
> > transcoding happens automatically.  I remember not setting this
> > variable in other installations and transcoding used to work out of
> > the box. Is media_mix_inbound_outbound_codecs=true default in
> > Freeswitch?
> >
> >> On 14 February 2016 at 13:56, Russell Treleaven
<rtreleaven at bunnykick.ca<mailto:rtreleaven at bunnykick.ca>> wrote:
> >> fyi
> >> https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiat
> >> ion
> >>
> >>> On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <rajil.s at gmail.com<mailto:rajil.s at gmail.com>>
wrote:
> >>>
> >>> The siptrace is at http://pastebin.com/xiGqtj1Y
> >>>
> >>> The call is being made from 303 (Android/CSipsimple with OPUS
> >>> codec) to 208 (pjsua test client with PCMU codec). The error is on
line 545.
> >>>
> >>> On 14 February 2016 at 11:28, Giovanni Maruzzelli
> >>> <gmaruzz at gmail.com<mailto:gmaruzz at gmail.com>>
> >>> wrote:
> >>>> How you originate the call? Is a bridge? >From which phone?
> >>>>
> >>>> Also, please pastebin the complete sip trace (from start of leg A
> >>>> to end of both legs) and put here a link to pastebin
> >>>>
> >>>> Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s at gmail.com<mailto:rajil.s at gmail.com>> ha
scritto:
> >>>>>
> >>>>> Hello,
> >>>>>
> >>>>> I have a remote sip phone (Linksys SPA3102) which only supports
PCMU.
> >>>>> When I call to this remote sip phone i get a 406 error that opus
> >>>>> is not supported as shown by the sip trace below. However, if I
> >>>>> force the codec to absolute like this
> >>>>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.<mailto:303 at 192.168.1.>
> >>>>> 5
> >>>>> the call works fine.
> >>>>>
> >>>>> Is there anyway I can make FreeSWITCH to automatically transcode
> >>>>> without forcing the codec string in the dial plan?
> >>>>>
> >>>>> The codec preferences is set as
> >>>>> global_codec_prefs=OPUS,PCMU,PCMA and
> >>>>> outbound_codec_prefs=PCMU,PCMA,GSM
> >>>>>
> >>>>> ---------------------------siptrace-----------------------------
> >>>>> ---
> >>>>>
> >>>>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
> >>>>>
> >>>>>
> >>>>>
------------------------------------------------------------------------
> >>>>>   SIP/2.0 406 Not Acceptable
> >>>>>   Via: SIP/2.0/UDP
> >>>>>
> >>>>>
> >>>>>
192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
> >>>>>   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
> >>>>>   From: "202" <sip:202 at 192.168.1.111<mailto:sip%3A202 at 192.168.1.111>>;tag=DFX0FUvr2vNcm
> >>>>>   To: <sip:303 at 192.168.1.5<mailto:sip%3A303 at 192.168.1.5>>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
> >>>>>   CSeq: 87372504 INVITE
> >>>>>   Content-Length:  0
> >>>>>
> >>>>>
> >>>>>
> >>>>> ----------------------------------------------------------------
> >>>>> -------- send 324 bytes to udp/[192.168.1.5]:5060 at
> >>>>> 08:02:16.368591:
> >>>>>
> >>>>>
> >>>>>
------------------------------------------------------------------------
> >>>>>   ACK sip:303 at 192.168.1.5<mailto:sip%3A303 at 192.168.1.5> SIP/2.0
> >>>>>   Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
> >>>>>   Max-Forwards: 68
> >>>>>   From: "202" <sip:202 at 192.168.1.111<mailto:sip%3A202 at 192.168.1.111>>;tag=DFX0FUvr2vNcm
> >>>>>   To: <sip:303 at 192.168.1.5<mailto:sip%3A303 at 192.168.1.5>>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
> >>>>>   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
> >>>>>   CSeq: 87372504 ACK
> >>>>>   Content-Length: 0
> >>>>>
> >>>>>
> >>>>>
> >>>>> ----------------------------------------------------------------
> >>>>> --------
> >>>>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
> >>>>> sofia/internal/303 at 192.168.1.5<mailto:303 at 192.168.1.5> entering state [terminated][406]
> >>>>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
> >>>>> sofia/internal/303 at 192.168.1.5<mailto:303 at 192.168.1.5> [CS_CONSUME_MEDIA]
> >>>>> [SERVICE_NOT_IMPLEMENTED]
> >>>>>
> >>>>> Thanks
> >>>>> Rajil
> >>>>>
> >>>>>
> >>>>> ________________________________________________________________
> >>>>> _________ 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/freeswit
> >>>>> ch-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/freeswitc
> >>>> h-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<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-u
> > sers
> > 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-use
> rs
> 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-use
> rs
> 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-use
> rs
> 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-use
> rs
> 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



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

? http://freeswitch.org/  ? http://cluecon.com/  ? http://twitter.com/FreeSWITCH
? irc.freenode.net<http://irc.freenode.net> #freeswitch ? http://freeswitch.org/g+
ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>  ? +19193869900

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From mgg at giagnocavo.net  Tue Feb 16 01:01:52 2016
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Mon, 15 Feb 2016 22:01:52 +0000
Subject: [Freeswitch-users] High availability on different networks
In-Reply-To: <1382DEBE-2DB3-48AC-AE54-5AD9EE370709@kavun.ch>
References: <78429801-FCA4-4BA9-8EA6-64A1BD7ECA28@kavun.ch>
	<CACOYK=gUYrGPouHq4voAygR7ctBOc6sdwBrAF8DWEeW6XkXS9A@mail.gmail.com>
	<CACOYK=jtfv356yOWXSJuqcVNL0L+_rMWAiQKEWqAC9i8bOwqUg@mail.gmail.com>
	<CAMbW9oJxHhWjcVvV7TdtzHMaxWGsu4L8kG-8YYmaNKXJPObgNQ@mail.gmail.com>
	<1382DEBE-2DB3-48AC-AE54-5AD9EE370709@kavun.ch>
Message-ID: <DM2PR07MB702AB76423288FA42DA3E9AD4AC0@DM2PR07MB702.namprd07.prod.outlook.com>

For outbound calls, you should be fine as long as your frontend SIP proxies can reach any backend (media/app/whatever) server. This is exactly what Via is for, right?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Emrah
Sent: Sunday, 14 February, 2016 10:18
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] High availability on different networks

I love all your ideas, and thanks for sharing.
The best option remains to test things out in real scenarios and see what happens.
I?m already distributing my media traffic. Now I?d like to make the SIP part redundant.
Just got some server resources in different locations. I?ll report back with my findings!

E

On Feb 12, 2016, at 10:09 AM, Andrew Cassidy <andrew at cassidywebservices.co.uk<mailto:andrew at cassidywebservices.co.uk>> wrote:

It's not instant, but I've used OVH failover IP's to do that sort of thing. Heartbeat over ipsec tunnel, use pacemaker with a custom script that does the OVH API call to move the IP address.

Sadly it's not that quick, takes about 2 minutes.

On 12 February 2016 at 07:40, Stanislav Sinyagin <ssinyagin at gmail.com<mailto:ssinyagin at gmail.com>> wrote:
there is an issue with anycast routing though: when you bring up a new
server, some running sessions will be dropped, because their IP
packets would be routed to a different host. So, it needs a careful
design. Maybe place only the SIP proxy on an anycast address, and run
the calls from unique local addresses.


Multiple DNS SRV records with different priorities are also possible,
but you can't direct the users to the nearest location within the same
domain. Also a bunch of SIP clients needs to be tested and you need to
answer the questions, like:

-- what is the timeout if the primary server is unavailable?
-- if the primary host goes down during the call, how soon can the
client re-dial?
-- what happens if the primary server comes up again?






On Thu, Feb 11, 2016 at 8:00 PM, Stanislav Sinyagin <ssinyagin at gmail.com<mailto:ssinyagin at gmail.com>> wrote:
> hi Emrah and all,
>
> it's the first time I actually searched for it, but there are hosting
> offers with anycast IP routing. It means, you have multiple servers in
> various locations, and they share the same service IP address. The
> clients connect to the nearest server, which is determined by standard
> BGP routing. You are still limited to a single global hosting
> provider, but you benefit from its redundant network and geographical
> distribution.
>
> In case of anycast addressing, incoming connections will be served
> easily. But the outgoing connections are rather tricky: you will need
> to bring the outbound call to the physical server where the user has
> registered, and initiate the connection from its anycast address. So,
> you can share and replicate the registration database, but you need to
> send the outbound call to the server which accepted the registration.
> I guess you should be able to retrieve this information from the
> registration database. This needs to be looked in details.
>
> Google for anycast server hosting, and there are at least 3 providers
> offering virtual hosts, and OVH is offering physical hosts as well. I
> guess there are more providers with similar offerings.
>
>
> Without anycast, you would need to use redundant registrars sharing
> the same service IP address -- for example, Digitalocean offers such
> service within any single datacenter.
>
> Having multiple registrars with different IP addresses is also
> possible, but then you depend on the way how each particular SIP
> client handles multiple IP addresses after resolving the domain name.
> Some of them may get stuck to a single address, even if it's not
> responding.
>
>
> cheers,
> stanislav
>
> On Thu, Feb 11, 2016 at 5:53 PM, Emrah <lists at kavun.ch<mailto:lists at kavun.ch>> wrote:
>> Hi list,
>> I?m writing to gather your thoughts and suggestions on how to have a high availability FS setup on different networks.
>>
>> I am trying to achieve the following:
>>  - Load balance FreeSWITCH instances on 2 or more servers, possibly in different countries.
>> - Shared user directory and dialplan, but I?m not sure if shared registrations would make sense.
>> - If a server goes down, the phone should register on the alternative servers. Obviously we can?t keep calls up.
>>
>> I?m obviously not the first one out there doing this. I?m trying to learn from those who?ve come up with reliable solutions.
>>
>> I?ve tried sharing a registration table among multiple FS instances. But it was a beginners mistake. Even with the right path to reach the client, only the invites sent from the server used by the phone would be processed.
>> If my phone registers on server A, then server A shares the info with server B, server B knows how to contact the phone but it won?t be able to. Supposedly because of NAT issues.
>>
>> I am aiming for fully independent FS instances that can back each other up and be used independently. I am guessing this would require some sort of SBC or external registrar server with a Kamailio or Repro.
>>
>> Anyway just trying to spark the conversation around this subject and hopefully we can come up with a formula that can help many with their FS deployments. My provider?s network just went all down in IPv4 and HA behind the same provider proved to be useless.
>>
>> Best,
>> E
>> _________________________________________________________________________
>> 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/>



--
Andrew Cassidy BSc (Hons) MBCS SSCA
Managing Director
[http://www.cassidywebservices.co.uk/static/emailsig.png]

T<mailto:info at cassidywebservices.co.uk> 03300 100 960  F<mailto:info at cassidywebservices.co.uk> 03300 100 961
E<mailto:info at cassidywebservices.co.uk> andrew at cassidywebservices.co.uk<mailto:andrew at cassidywebservices.co.uk>
W<mailto:info at cassidywebservices.co.uk> www.cassidywebservices.co.uk<http://www.cassidywebservices.co.uk/>
_________________________________________________________________________
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 lists at telefaks.de  Tue Feb 16 01:36:43 2016
From: lists at telefaks.de (Peter Steinbach)
Date: Mon, 15 Feb 2016 23:36:43 +0100
Subject: [Freeswitch-users] multiple internal ip
In-Reply-To: <857665560.7491164.1455569014252.JavaMail.yahoo@mail.yahoo.com>
References: <857665560.7491164.1455569014252.JavaMail.yahoo.ref@mail.yahoo.com>
	<857665560.7491164.1455569014252.JavaMail.yahoo@mail.yahoo.com>
Message-ID: <56C252FB.6060808@telefaks.de>

We have build a system with several IPs including multiple VPNs. For
this purpose, we created a number of profiles, one for each IP and
grouped them to contextes.
This is possible, but you neeed to take care about your dialstrings when
dialling to phones in distinct profiles. This may be more difficult with
static xml configs, so with xml-curl it can be handled in a better way.

As Brian commented, a multi tenant setup can be setup with distinct
domains. But there may be of course other reasons for using distinct IPs.
 
Bonne chance!

Peter


On 02/15/16 21:43, Antoine Durant wrote:
> Hi,
>
> I would like to have more internal IP addresses to do some "Multiple
> Companies".
>
> The WAN IP is used (80.X.X.125) to mount all trunk (1 trunk by
> company) to the same provider sip.
>
> Is it possible to have an internal IP for each company:
> company-a.org => 192.168.1.1
> company-b.org => 192.168.1.2
> company-c.org => 192.168.1.3
> company-X.org => 192.168.1.X
>
> How to use FreeSwitch this way?
>
> 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


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de

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From ssinyagin at gmail.com  Tue Feb 16 01:43:46 2016
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Mon, 15 Feb 2016 23:43:46 +0100
Subject: [Freeswitch-users] multiple internal ip
In-Reply-To: <857665560.7491164.1455569014252.JavaMail.yahoo@mail.yahoo.com>
References: <857665560.7491164.1455569014252.JavaMail.yahoo.ref@mail.yahoo.com>
	<857665560.7491164.1455569014252.JavaMail.yahoo@mail.yahoo.com>
Message-ID: <CACOYK=ixdn9vpMvAWAd_yUv7Xa0OSz7E469igSxnymB2GpmJtA@mail.gmail.com>

each SIP profile binds to a unique combination of IP address and
TCP+UDP port. So, you can define as many SIP profiles as you need, and
route the inbound calls into different contexts as needed.

But as other colleagues explained, distinctive DNS domains are usually
sufficient for a multi-tenancy setup.

I outlined some details in this article:
https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+PBX+Example





On Mon, Feb 15, 2016 at 9:43 PM, Antoine Durant
<antoine.durant29 at yahoo.fr> wrote:
> Hi,
>
> I would like to have more internal IP addresses to do some "Multiple
> Companies".
>
> The WAN IP is used (80.X.X.125) to mount all trunk (1 trunk by company) to
> the same provider sip.
>
> Is it possible to have an internal IP for each company:
> company-a.org => 192.168.1.1
> company-b.org => 192.168.1.2
> company-c.org => 192.168.1.3
> company-X.org => 192.168.1.X
>
> How to use FreeSwitch this way?
>
> 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 rajil.s at gmail.com  Tue Feb 16 01:44:53 2016
From: rajil.s at gmail.com (Rajil Saraswat)
Date: Mon, 15 Feb 2016 16:44:53 -0600
Subject: [Freeswitch-users] Where to put Jitterbuffer?
Message-ID: <CAFoaQoDpV-a478WDkPcFTAMO+zxg4Nx8bk9B9gHtK5dZSgrs-g@mail.gmail.com>

Hello,

I have two FS instances connected to each other over VPN. The VPN
sometimes has poor connectivity and I was wondering whether
jitterbuffer can help to improve the call quality. I am using the OPUS
codec between the FS instances since it is supposed to behave better
in lower bandwidth. The ATA on both ends are Linksys SPA3102.

My connections is as follows:

ATA_A---->FS_A---->vpn---->FS_B---->ATA_B


When user A is calling to B where should i specify jitter buffer
amongst the following:

1. The bridge between Freeswitch instance FS_A and FS_B
<action application="bridge"
data="{jitterbuffer_msec=60}sofia/gateway/$1 at FS_B_IP"/>

2. The bridge between Freeswitch instance FS_B and ATA_B
<action application="bridge"
data="{jitterbuffer_msec=60}sofia/internal/$1 at ATA_B_IP"/>

Thanks,
Rajil


From andretodd at verizon.net  Tue Feb 16 02:03:45 2016
From: andretodd at verizon.net (Andre DeMattia)
Date: Mon, 15 Feb 2016 18:03:45 -0500
Subject: [Freeswitch-users] SWITCH_EVENT_CHANNEL_PROGRESS_MEDIA
Message-ID: <061201d16845$1f679810$5e36c830$@verizon.net>

Hi is there an application that will give me the same results as this event?
switch_event_types_t.SWITCH_EVENT_CHANNEL_PROGRESS_MEDIA I see
<https://wiki.freeswitch.org/wiki/Variable_execute_on_ring> execute_on_ring
and execute_on_media. Wondering if either of these are the same as
switch_event_types_t.SWITCH_EVENT_CHANNEL_PROGRESS_MEDIA

 

Thanks

Andre

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From anthony.minessale at gmail.com  Tue Feb 16 02:06:35 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Feb 2016 17:06:35 -0600
Subject: [Freeswitch-users] SWITCH_EVENT_CHANNEL_PROGRESS_MEDIA
In-Reply-To: <061201d16845$1f679810$5e36c830$@verizon.net>
References: <061201d16845$1f679810$5e36c830$@verizon.net>
Message-ID: <CAKbxfG9RKKo0YHbk39xtQnkkdNC2RvxovYNY8+Jh1kuyz_th5A@mail.gmail.com>

execute_on_pre_answer

On Mon, Feb 15, 2016 at 5:03 PM, Andre DeMattia <andretodd at verizon.net>
wrote:

> Hi is there an application that will give me the same results as this
> event? switch_event_types_t.SWITCH_EVENT_CHANNEL_PROGRESS_MEDIA I see
> execute_on_ring
> <https://wiki.freeswitch.org/wiki/Variable_execute_on_ring> and
> execute_on_media. Wondering if either of these are the same as
> switch_event_types_t.SWITCH_EVENT_CHANNEL_PROGRESS_MEDIA
>
>
>
> Thanks
>
> Andre
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Tue Feb 16 02:10:11 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Feb 2016 17:10:11 -0600
Subject: [Freeswitch-users] Where to put Jitterbuffer?
In-Reply-To: <CAFoaQoDpV-a478WDkPcFTAMO+zxg4Nx8bk9B9gHtK5dZSgrs-g@mail.gmail.com>
References: <CAFoaQoDpV-a478WDkPcFTAMO+zxg4Nx8bk9B9gHtK5dZSgrs-g@mail.gmail.com>
Message-ID: <CAKbxfG8_9P_y7dm5cf1pcxphEOkztO=5=E4aUHETXwQLdw9xXQ@mail.gmail.com>

The variable must be set on the leg before the channel engages media.
you can test if its on with

uuid_jitterbuffer <uuid> debug:10


if you want it on both legs set the following global vars in vars.xml

rtp_jitter_buffer_during_bridge=true

jitterbuffer_msec=60






On Mon, Feb 15, 2016 at 4:44 PM, Rajil Saraswat <rajil.s at gmail.com> wrote:

> Hello,
>
> I have two FS instances connected to each other over VPN. The VPN
> sometimes has poor connectivity and I was wondering whether
> jitterbuffer can help to improve the call quality. I am using the OPUS
> codec between the FS instances since it is supposed to behave better
> in lower bandwidth. The ATA on both ends are Linksys SPA3102.
>
> My connections is as follows:
>
> ATA_A---->FS_A---->vpn---->FS_B---->ATA_B
>
>
> When user A is calling to B where should i specify jitter buffer
> amongst the following:
>
> 1. The bridge between Freeswitch instance FS_A and FS_B
> <action application="bridge"
> data="{jitterbuffer_msec=60}sofia/gateway/$1 at FS_B_IP"/>
>
> 2. The bridge between Freeswitch instance FS_B and ATA_B
> <action application="bridge"
> data="{jitterbuffer_msec=60}sofia/internal/$1 at ATA_B_IP"/>
>
> Thanks,
> Rajil
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 brian at freeswitch.org  Tue Feb 16 02:48:19 2016
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Feb 2016 17:48:19 -0600
Subject: [Freeswitch-users] multiple internal ip
In-Reply-To: <CACOYK=ixdn9vpMvAWAd_yUv7Xa0OSz7E469igSxnymB2GpmJtA@mail.gmail.com>
References: <857665560.7491164.1455569014252.JavaMail.yahoo.ref@mail.yahoo.com>
	<857665560.7491164.1455569014252.JavaMail.yahoo@mail.yahoo.com>
	<CACOYK=ixdn9vpMvAWAd_yUv7Xa0OSz7E469igSxnymB2GpmJtA@mail.gmail.com>
Message-ID: <CAEJMVkDqf7ijKtq7A_+c2UTS4=zDzhm3HY=8g0tLvaTkqV4Agw@mail.gmail.com>

You can if you want put multiple domains on a single profile... just sayin!
:)

On Mon, Feb 15, 2016 at 4:43 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> each SIP profile binds to a unique combination of IP address and
> TCP+UDP port. So, you can define as many SIP profiles as you need, and
> route the inbound calls into different contexts as needed.
>
> But as other colleagues explained, distinctive DNS domains are usually
> sufficient for a multi-tenancy setup.
>
> I outlined some details in this article:
> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+PBX+Example
>
>
>
>
>
> On Mon, Feb 15, 2016 at 9:43 PM, Antoine Durant
> <antoine.durant29 at yahoo.fr> wrote:
> > Hi,
> >
> > I would like to have more internal IP addresses to do some "Multiple
> > Companies".
> >
> > The WAN IP is used (80.X.X.125) to mount all trunk (1 trunk by company)
> to
> > the same provider sip.
> >
> > Is it possible to have an internal IP for each company:
> > company-a.org => 192.168.1.1
> > company-b.org => 192.168.1.2
> > company-c.org => 192.168.1.3
> > company-X.org => 192.168.1.X
> >
> > How to use FreeSwitch this way?
> >
> > 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
>



-- 

*Brian West*
brian at freeswitch.org


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

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 rajil.s at gmail.com  Tue Feb 16 02:54:54 2016
From: rajil.s at gmail.com (Rajil Saraswat)
Date: Mon, 15 Feb 2016 17:54:54 -0600
Subject: [Freeswitch-users] Where to put Jitterbuffer?
In-Reply-To: <CAKbxfG8_9P_y7dm5cf1pcxphEOkztO=5=E4aUHETXwQLdw9xXQ@mail.gmail.com>
References: <CAFoaQoDpV-a478WDkPcFTAMO+zxg4Nx8bk9B9gHtK5dZSgrs-g@mail.gmail.com>
	<CAKbxfG8_9P_y7dm5cf1pcxphEOkztO=5=E4aUHETXwQLdw9xXQ@mail.gmail.com>
Message-ID: <CAFoaQoDXyKhNGury7Lu33auLcJ4k2ya3iSXx2N-vKaKgxgnAWg@mail.gmail.com>

The wiki page (https://wiki.freeswitch.org/wiki/Jitterbuffer) says,

If both sides of a bridge are RTP and both sides have a jb, its fairly
useless.  In fact if anything, it can worsen call quality.

So if I set jitterbuffer on both of my FS instances as global
variables, wont that be against what the wiki says?


On 15 February 2016 at 17:10, Anthony Minessale <anthony.minessale at gmail.com
> wrote:

> The variable must be set on the leg before the channel engages media.
> you can test if its on with
>
> uuid_jitterbuffer <uuid> debug:10
>
>
> if you want it on both legs set the following global vars in vars.xml
>
> rtp_jitter_buffer_during_bridge=true
>
> jitterbuffer_msec=60
>
>
>
>
>
>
> On Mon, Feb 15, 2016 at 4:44 PM, Rajil Saraswat <rajil.s at gmail.com> wrote:
>
>> Hello,
>>
>> I have two FS instances connected to each other over VPN. The VPN
>> sometimes has poor connectivity and I was wondering whether
>> jitterbuffer can help to improve the call quality. I am using the OPUS
>> codec between the FS instances since it is supposed to behave better
>> in lower bandwidth. The ATA on both ends are Linksys SPA3102.
>>
>> My connections is as follows:
>>
>> ATA_A---->FS_A---->vpn---->FS_B---->ATA_B
>>
>>
>> When user A is calling to B where should i specify jitter buffer
>> amongst the following:
>>
>> 1. The bridge between Freeswitch instance FS_A and FS_B
>> <action application="bridge"
>> data="{jitterbuffer_msec=60}sofia/gateway/$1 at FS_B_IP"/>
>>
>> 2. The bridge between Freeswitch instance FS_B and ATA_B
>> <action application="bridge"
>> data="{jitterbuffer_msec=60}sofia/internal/$1 at ATA_B_IP"/>
>>
>> Thanks,
>> Rajil
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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  Tue Feb 16 03:23:01 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Feb 2016 18:23:01 -0600
Subject: [Freeswitch-users] Where to put Jitterbuffer?
In-Reply-To: <CAFoaQoDXyKhNGury7Lu33auLcJ4k2ya3iSXx2N-vKaKgxgnAWg@mail.gmail.com>
References: <CAFoaQoDpV-a478WDkPcFTAMO+zxg4Nx8bk9B9gHtK5dZSgrs-g@mail.gmail.com>
	<CAKbxfG8_9P_y7dm5cf1pcxphEOkztO=5=E4aUHETXwQLdw9xXQ@mail.gmail.com>
	<CAFoaQoDXyKhNGury7Lu33auLcJ4k2ya3iSXx2N-vKaKgxgnAWg@mail.gmail.com>
Message-ID: <CAKbxfG-tjByKXE4rb=jDjma2+nJJ7hZb=q2frNunnzdCKeaLJQ@mail.gmail.com>

If one or the other end of the call does not have its own JB then putting
it on both sides can fix the stream for the other end.
That is why its not defaulted to being possible but you asked how to do it.


On Mon, Feb 15, 2016 at 5:54 PM, Rajil Saraswat <rajil.s at gmail.com> wrote:

>
> The wiki page (https://wiki.freeswitch.org/wiki/Jitterbuffer) says,
>
> If both sides of a bridge are RTP and both sides have a jb, its fairly
> useless.  In fact if anything, it can worsen call quality.
>
> So if I set jitterbuffer on both of my FS instances as global variables, wont that be against what the wiki says?
>
>
> On 15 February 2016 at 17:10, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> The variable must be set on the leg before the channel engages media.
>> you can test if its on with
>>
>> uuid_jitterbuffer <uuid> debug:10
>>
>>
>> if you want it on both legs set the following global vars in vars.xml
>>
>> rtp_jitter_buffer_during_bridge=true
>>
>> jitterbuffer_msec=60
>>
>>
>>
>>
>>
>>
>> On Mon, Feb 15, 2016 at 4:44 PM, Rajil Saraswat <rajil.s at gmail.com>
>> wrote:
>>
>>> Hello,
>>>
>>> I have two FS instances connected to each other over VPN. The VPN
>>> sometimes has poor connectivity and I was wondering whether
>>> jitterbuffer can help to improve the call quality. I am using the OPUS
>>> codec between the FS instances since it is supposed to behave better
>>> in lower bandwidth. The ATA on both ends are Linksys SPA3102.
>>>
>>> My connections is as follows:
>>>
>>> ATA_A---->FS_A---->vpn---->FS_B---->ATA_B
>>>
>>>
>>> When user A is calling to B where should i specify jitter buffer
>>> amongst the following:
>>>
>>> 1. The bridge between Freeswitch instance FS_A and FS_B
>>> <action application="bridge"
>>> data="{jitterbuffer_msec=60}sofia/gateway/$1 at FS_B_IP"/>
>>>
>>> 2. The bridge between Freeswitch instance FS_B and ATA_B
>>> <action application="bridge"
>>> data="{jitterbuffer_msec=60}sofia/internal/$1 at ATA_B_IP"/>
>>>
>>> Thanks,
>>> Rajil
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 rajil.s at gmail.com  Tue Feb 16 03:33:59 2016
From: rajil.s at gmail.com (Rajil Saraswat)
Date: Mon, 15 Feb 2016 18:33:59 -0600
Subject: [Freeswitch-users] Where to put Jitterbuffer?
In-Reply-To: <CAKbxfG-tjByKXE4rb=jDjma2+nJJ7hZb=q2frNunnzdCKeaLJQ@mail.gmail.com>
References: <CAFoaQoDpV-a478WDkPcFTAMO+zxg4Nx8bk9B9gHtK5dZSgrs-g@mail.gmail.com>
	<CAKbxfG8_9P_y7dm5cf1pcxphEOkztO=5=E4aUHETXwQLdw9xXQ@mail.gmail.com>
	<CAFoaQoDXyKhNGury7Lu33auLcJ4k2ya3iSXx2N-vKaKgxgnAWg@mail.gmail.com>
	<CAKbxfG-tjByKXE4rb=jDjma2+nJJ7hZb=q2frNunnzdCKeaLJQ@mail.gmail.com>
Message-ID: <CAFoaQoC9TdrkssheVEE9Pux_5-Bh27b4nJtOou5FjbgDGzZaOA@mail.gmail.com>

Ok, what about my specific use case where both endpoints ATA (Linksys
spa3102) do have internal jitterbuffer. Should i still set the global
variables as you suggested to improve the call quality over vpn?

On 15 February 2016 at 18:23, Anthony Minessale <anthony.minessale at gmail.com
> wrote:

> If one or the other end of the call does not have its own JB then putting
> it on both sides can fix the stream for the other end.
> That is why its not defaulted to being possible but you asked how to do it.
>
>
> On Mon, Feb 15, 2016 at 5:54 PM, Rajil Saraswat <rajil.s at gmail.com> wrote:
>
>>
>> The wiki page (https://wiki.freeswitch.org/wiki/Jitterbuffer) says,
>>
>> If both sides of a bridge are RTP and both sides have a jb, its fairly
>> useless.  In fact if anything, it can worsen call quality.
>>
>> So if I set jitterbuffer on both of my FS instances as global variables, wont that be against what the wiki says?
>>
>>
>> On 15 February 2016 at 17:10, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> The variable must be set on the leg before the channel engages media.
>>> you can test if its on with
>>>
>>> uuid_jitterbuffer <uuid> debug:10
>>>
>>>
>>> if you want it on both legs set the following global vars in vars.xml
>>>
>>> rtp_jitter_buffer_during_bridge=true
>>>
>>> jitterbuffer_msec=60
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Mon, Feb 15, 2016 at 4:44 PM, Rajil Saraswat <rajil.s at gmail.com>
>>> wrote:
>>>
>>>> Hello,
>>>>
>>>> I have two FS instances connected to each other over VPN. The VPN
>>>> sometimes has poor connectivity and I was wondering whether
>>>> jitterbuffer can help to improve the call quality. I am using the OPUS
>>>> codec between the FS instances since it is supposed to behave better
>>>> in lower bandwidth. The ATA on both ends are Linksys SPA3102.
>>>>
>>>> My connections is as follows:
>>>>
>>>> ATA_A---->FS_A---->vpn---->FS_B---->ATA_B
>>>>
>>>>
>>>> When user A is calling to B where should i specify jitter buffer
>>>> amongst the following:
>>>>
>>>> 1. The bridge between Freeswitch instance FS_A and FS_B
>>>> <action application="bridge"
>>>> data="{jitterbuffer_msec=60}sofia/gateway/$1 at FS_B_IP"/>
>>>>
>>>> 2. The bridge between Freeswitch instance FS_B and ATA_B
>>>> <action application="bridge"
>>>> data="{jitterbuffer_msec=60}sofia/internal/$1 at ATA_B_IP"/>
>>>>
>>>> Thanks,
>>>> Rajil
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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  Tue Feb 16 03:46:01 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Feb 2016 18:46:01 -0600
Subject: [Freeswitch-users] Where to put Jitterbuffer?
In-Reply-To: <CAFoaQoC9TdrkssheVEE9Pux_5-Bh27b4nJtOou5FjbgDGzZaOA@mail.gmail.com>
References: <CAFoaQoDpV-a478WDkPcFTAMO+zxg4Nx8bk9B9gHtK5dZSgrs-g@mail.gmail.com>
	<CAKbxfG8_9P_y7dm5cf1pcxphEOkztO=5=E4aUHETXwQLdw9xXQ@mail.gmail.com>
	<CAFoaQoDXyKhNGury7Lu33auLcJ4k2ya3iSXx2N-vKaKgxgnAWg@mail.gmail.com>
	<CAKbxfG-tjByKXE4rb=jDjma2+nJJ7hZb=q2frNunnzdCKeaLJQ@mail.gmail.com>
	<CAFoaQoC9TdrkssheVEE9Pux_5-Bh27b4nJtOou5FjbgDGzZaOA@mail.gmail.com>
Message-ID: <CAKbxfG-cRVrYLbqRs=xHEYFnuKVapTorw451dpy7d7+0iOb7Dg@mail.gmail.com>

When you keep the codec the same across the whole call path then the
jitterbuffer is not as important because FS will preserve the timestamps.
If you are transcoding or recording the call, then the JB can help because
you are creating new timestamps so the far end can no longer perceive the
jitter.

Ideally you should just try it.
Also you should make sure your vpn is using udp or gprs not tcp

You should probably try it with everything PCMU or PCMA as well with and
without the JB.
Trial and error is your friend.  But without the vars I mentioned it will
never use the JB during a bridge.






On Mon, Feb 15, 2016 at 6:33 PM, Rajil Saraswat <rajil.s at gmail.com> wrote:

> Ok, what about my specific use case where both endpoints ATA (Linksys
> spa3102) do have internal jitterbuffer. Should i still set the global
> variables as you suggested to improve the call quality over vpn?
>
> On 15 February 2016 at 18:23, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> If one or the other end of the call does not have its own JB then putting
>> it on both sides can fix the stream for the other end.
>> That is why its not defaulted to being possible but you asked how to do
>> it.
>>
>>
>> On Mon, Feb 15, 2016 at 5:54 PM, Rajil Saraswat <rajil.s at gmail.com>
>> wrote:
>>
>>>
>>> The wiki page (https://wiki.freeswitch.org/wiki/Jitterbuffer) says,
>>>
>>> If both sides of a bridge are RTP and both sides have a jb, its fairly
>>> useless.  In fact if anything, it can worsen call quality.
>>>
>>> So if I set jitterbuffer on both of my FS instances as global variables, wont that be against what the wiki says?
>>>
>>>
>>> On 15 February 2016 at 17:10, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> The variable must be set on the leg before the channel engages media.
>>>> you can test if its on with
>>>>
>>>> uuid_jitterbuffer <uuid> debug:10
>>>>
>>>>
>>>> if you want it on both legs set the following global vars in vars.xml
>>>>
>>>> rtp_jitter_buffer_during_bridge=true
>>>>
>>>> jitterbuffer_msec=60
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Mon, Feb 15, 2016 at 4:44 PM, Rajil Saraswat <rajil.s at gmail.com>
>>>> wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>> I have two FS instances connected to each other over VPN. The VPN
>>>>> sometimes has poor connectivity and I was wondering whether
>>>>> jitterbuffer can help to improve the call quality. I am using the OPUS
>>>>> codec between the FS instances since it is supposed to behave better
>>>>> in lower bandwidth. The ATA on both ends are Linksys SPA3102.
>>>>>
>>>>> My connections is as follows:
>>>>>
>>>>> ATA_A---->FS_A---->vpn---->FS_B---->ATA_B
>>>>>
>>>>>
>>>>> When user A is calling to B where should i specify jitter buffer
>>>>> amongst the following:
>>>>>
>>>>> 1. The bridge between Freeswitch instance FS_A and FS_B
>>>>> <action application="bridge"
>>>>> data="{jitterbuffer_msec=60}sofia/gateway/$1 at FS_B_IP"/>
>>>>>
>>>>> 2. The bridge between Freeswitch instance FS_B and ATA_B
>>>>> <action application="bridge"
>>>>> data="{jitterbuffer_msec=60}sofia/internal/$1 at ATA_B_IP"/>
>>>>>
>>>>> Thanks,
>>>>> Rajil
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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 mandra at gmail.com  Tue Feb 16 07:20:44 2016
From: mandra at gmail.com (Chris Mandra)
Date: Mon, 15 Feb 2016 23:20:44 -0500
Subject: [Freeswitch-users] bridging out and custom signaling question
In-Reply-To: <CAJdRzN5fJzDV-e7o6_sj4cy_Z-D8W+EG=qKpkkg33PyL0+7Udw@mail.gmail.com>
References: <CAJdRzN5fJzDV-e7o6_sj4cy_Z-D8W+EG=qKpkkg33PyL0+7Udw@mail.gmail.com>
Message-ID: <CAJdRzN6Nk7f8F1u_L1j=-oAfycHdWO61QqzW6ohikSBLdCW=KA@mail.gmail.com>

Any ideas on this?
On Mon, Feb 15, 2016 at 11:16 AM, Chris Mandra <mandra at gmail.com
<javascript:_e(%7B%7D,'cvml','mandra at gmail.com');>> wrote:

> Hi guys - I hope things are good where you are. I have a question:
>
> What is the easiest way to bridge a call out of freeswitch not using sip
> but using a custom signaling protocol that may be a restful protocol?
>
> Thank you !
>
> chris
>
>


-- 
mandra
c:410.258.5281


-- 
mandra
c:410.258.5281
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From krice at freeswitch.org  Tue Feb 16 07:55:34 2016
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 15 Feb 2016 22:55:34 -0600
Subject: [Freeswitch-users] bridging out and custom signaling question
In-Reply-To: <CAJdRzN6Nk7f8F1u_L1j=-oAfycHdWO61QqzW6ohikSBLdCW=KA@mail.gmail.com>
References: <CAJdRzN5fJzDV-e7o6_sj4cy_Z-D8W+EG=qKpkkg33PyL0+7Udw@mail.gmail.com>
	<CAJdRzN6Nk7f8F1u_L1j=-oAfycHdWO61QqzW6ohikSBLdCW=KA@mail.gmail.com>
Message-ID: <8DB037F6-0515-41A5-B265-7C3684B68433@freeswitch.org>

You'll have to write (or have written) an endpoint driver for it...  

Sent from my iPhone

> On Feb 15, 2016, at 10:20 PM, Chris Mandra <mandra at gmail.com> wrote:
> 
> Any ideas on this?
>> On Mon, Feb 15, 2016 at 11:16 AM, Chris Mandra <mandra at gmail.com> wrote:
>> Hi guys - I hope things are good where you are. I have a question: 
>> 
>> What is the easiest way to bridge a call out of freeswitch not using sip but using a custom signaling protocol that may be a restful protocol?
>> 
>> Thank you !
>> 
>> chris
>> 
> 
> 
> 
> -- 
> mandra
> c:410.258.5281
> 
> 
> -- 
> 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
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From gmaruzz at gmail.com  Tue Feb 16 10:57:44 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 16 Feb 2016 08:57:44 +0100
Subject: [Freeswitch-users] bridging out and custom signaling question
In-Reply-To: <8DB037F6-0515-41A5-B265-7C3684B68433@freeswitch.org>
References: <CAJdRzN5fJzDV-e7o6_sj4cy_Z-D8W+EG=qKpkkg33PyL0+7Udw@mail.gmail.com>
	<CAJdRzN6Nk7f8F1u_L1j=-oAfycHdWO61QqzW6ohikSBLdCW=KA@mail.gmail.com>
	<8DB037F6-0515-41A5-B265-7C3684B68433@freeswitch.org>
Message-ID: <CALXCt0p7rAM=SxWxKnWmmqNx0t5D=wNN6R8UWXHLGXKZpndrgg@mail.gmail.com>

You can have a look at mod-alsa (is very easy) and search into
http://www.freeswitch.org/confluence for info about writing endpoints in C.

Also, you can write at consulting at freeswitch.org for official, professional
FreeSWITCH development and deployment.
Il 16/Feb/2016 05:56, "Ken Rice" <krice at freeswitch.org> ha scritto:

> You'll have to write (or have written) an endpoint driver for it...
>
> Sent from my iPhone
>
> On Feb 15, 2016, at 10:20 PM, Chris Mandra <mandra at gmail.com> wrote:
>
> Any ideas on this?
> On Mon, Feb 15, 2016 at 11:16 AM, Chris Mandra <mandra at gmail.com> wrote:
>
>> Hi guys - I hope things are good where you are. I have a question:
>>
>> What is the easiest way to bridge a call out of freeswitch not using sip
>> but using a custom signaling protocol that may be a restful protocol?
>>
>> Thank you !
>>
>> chris
>>
>>
>
>
> --
> mandra
> c:410.258.5281
>
>
> --
> 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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 stefano.favaro at edistar.com  Tue Feb 16 11:40:43 2016
From: stefano.favaro at edistar.com (Stefano Favaro)
Date: Tue, 16 Feb 2016 09:40:43 +0100 (CET)
Subject: [Freeswitch-users] Problem with mod_spy
In-Reply-To: <CABMX_w-yfbfctKd2bF=p9E-VB0yi9vV952Z8-dbwcg22FEou2g@mail.gmail.com>
Message-ID: <775731365.7256.1455612043715.JavaMail.root@mailserver.edistar.com>

Thanks, 

I've tried with 


<extension name="eavesdrop"> 
<condition field="destination_number" expression="^88(.*)$|^\*0(.*)$"> 
<action application="answer"/> 
<action application="log" data="$1" /> 
<action application="userspy" data="user/$1"/> 
</condition> 
</extension> 

but unfortunately I have the same problem. 

SF. 

https://pastebin.freeswitch.org/24556 


----- Messaggio originale -----
Da: "?talo Rossi" <italo at freeswitch.org> 
A: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org> 
Inviato: Luned?, 15 febbraio 2016 20:57:29 
Oggetto: Re: [Freeswitch-users] Problem with mod_spy 


It's probably due to the way you're dialing to your user, I can see in your log this bridge command: bridge(sofia/internal/4750%myserver) 


Just try to dial bridge user/4750 it'll work if 4750 is registered with FS. 


On Mon, Feb 15, 2016 at 6:12 AM, Stefano Favaro < stefano.favaro at edistar.com > wrote: 




Thank you Italo. 

This is my log 

https://pastebin.freeswitch.org/24555 

and this is the id of my user spy call 

94bae5a0-dd2c-4d0e-99fd-8ef81e753227 

Stefano 


Da: "?talo Rossi" < italo at freeswitch.org > 
A: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org > 
Inviato: Gioved?, 11 febbraio 2016 18:52:46 
Oggetto: Re: [Freeswitch-users] Problem with mod_spy 




Stefano, 


Can you post your debug logs (/log 7)? 


Use https://pastebin.freeswitch.org/ 


On Thu, Feb 11, 2016 at 1:33 PM, Stefano Favaro < stefano.favaro at edistar.com > wrote: 




Hello,

 I have a problem with the mod_spy module. 
It seems that it just plays music and do not actually spy.
I want to dial an extension with the number of the user I want to spy and wait for a call in or out for that user. 
Mod_spy application plays music but when a call is handled by the user I can't hear it continues to play the music on hold audio file. 

This is the dialplan:

<extension name="eavesdrop"> 
<condition field="destination_number" expression="^88(.*)$|^\*0(.*)$"> 
<action application="answer"/> 
<action application="userspy" data="$1 at myserver"/> 
</condition> 
</extension> 
If I dial 881000, for example, It means I want to spy on user 1000.
I have in and out calls from user 1000 but I can't hear. 
userspy_show in fs_cli, I get : 
1000 at myserver : e9165eaf-e8d7-40ad-a8cd-d2963f363684

1 total spy 

I have FreeSwitch version FreeSWITCH Version 1.4.26-37~64bit (-37 64bit) 

SF. 

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

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

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

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

FreeSWITCH-users mailing list 
FreeSWITCH-users at 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 pskoul at gmail.com  Tue Feb 16 14:35:38 2016
From: pskoul at gmail.com (Panagiotis Skoulikaritis)
Date: Tue, 16 Feb 2016 13:35:38 +0200
Subject: [Freeswitch-users] extra header account code is not written to
 cdr if cancel is received a few ms after invite
In-Reply-To: <56B7A25A.8090801@gmail.com>
References: <56AE42E1.3080703@gmail.com>
	<3CBD15A5-819B-487B-9DE0-C8120DDCAC94@gmail.com>
	<3D075DBF-8FEF-40CE-9922-A7CFC462512E@gmail.com>
	<56B7A25A.8090801@gmail.com>
Message-ID: <56C3098A.1040000@gmail.com>

What possibly could cause the account code to not be written in the cdr?
I have seen from traces that it is present on the sip mesage,
Unfortunately the loging is disabled I will enable it to see if I can 
see anything.

But from what I see is that it doesn't write the account code on the cdr 
when the outgoing leg for whatever reason is not made.

any advice ?

Regards

Panagiotis

On 7/2/2016 10:00 ??, Panagiotis Skoulikaritis wrote:
> I have tried both inline and export but I still have cdrs where the
> accountcode is not written.
>
> Any help would be greatly appreciated.
>
> Regards
>
> Panagiotis
>
>
> On 1/31/2016 11:58 PM, Oz Mortimer wrote:
>> Try export rather than set
>>
>>> On 31 Jan 2016, at 18:45, servtelar at gmail.com wrote:
>>>
>>> Shouldn't that be done as inline?
>>>
>>> Sent from my iPhone
>>>
>>>> On Jan 31, 2016, at 12:22 PM, Panagiotis Skoulikaritis <pskoul at gmail.com> wrote:
>>>>
>>>> Dear all
>>>>
>>>> I have an implementation FreeSWITCH as a sort of SBC, it is used to send
>>>> the calls to the terminating carriers and do topology hiding, nothing
>>>> fancy. Also I gather cdrs from the FreeSWITCH.
>>>>
>>>> In order to distinguish each customer on the FS cdrs I send an extra
>>>> header containing the accountcode.
>>>>
>>>> I have noticed that if the call is canceled immediately on the same sec,
>>>> the account code is not written on the cdr.
>>>> To be more precise the cancel is send a few milliseconds after it has
>>>> received the invite, and before the FreeSWITCH has sent the call to the
>>>> terminating carrier (I'm using Homer Sipcapture to capture all the
>>>> traces and I don't see an attempt being made at the terminating carrier)
>>>> also I don't see a b-leg cdr.
>>>>
>>>> FreeSWITCH is writing both a-leg and b-leg cdrs in csv format.
>>>>
>>>> The dialplan that I use is simple
>>>>
>>>> <extension name="carrier-1">
>>>> <condition field="network_addr"
>>>> expression="^(^xx\.xx\.xx\.xx|^yy\.yy\.yy\.yy)$"/>
>>>> <condition field="destination_number" expression="^2#(\d+)$">
>>>> <action application="set" data="call_direction=outgoing"/>
>>>> <action application="set" data="hangup_after_bridge=true"/>
>>>> <action application="set" data="continue_on_fail=true"/>
>>>> <action application="set" data="sip_cid_in_1xx=false"/>
>>>> <action application="set" data="inherit_codec=true"/>
>>>> <action application="set" data="accountcode=${sip_h_X-ACCOUNTCODE}"/>
>>>> <action application="unset" data="sip_h_X-ACCOUNTCODE"/>
>>>> <action application="set" data="execute_on_answer=sched_hangup +21600
>>>> alloted_timeout"/>
>>>> <action application="bridge" data="sofia/external/4567$1 at zz.zz.zz.zz:5070"/>
>>>> </condition>
>>>> </extension>
>>>>
>>>>
>>>> any idea how I can make sure that the account code will always be written ?
>>>>
>>>>
>>>> Best Regards
>>>>
>>>> Panagiotis
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 italo at freeswitch.org  Tue Feb 16 15:47:54 2016
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Tue, 16 Feb 2016 09:47:54 -0300
Subject: [Freeswitch-users] Problem with mod_spy
In-Reply-To: <775731365.7256.1455612043715.JavaMail.root@mailserver.edistar.com>
References: <CABMX_w-yfbfctKd2bF=p9E-VB0yi9vV952Z8-dbwcg22FEou2g@mail.gmail.com>
	<775731365.7256.1455612043715.JavaMail.root@mailserver.edistar.com>
Message-ID: <CABMX_w_SGWBLR9g3Y0skAKnkWprdM4kK3aypGGpfdk51DUHp3w@mail.gmail.com>

I mean your bridge string, not userspy, instead of sofia/internal/4750 at ...
just use bridge(user/4750) to DIAL to that user. Your userspy args looks
good

On Tue, Feb 16, 2016 at 5:40 AM, Stefano Favaro <stefano.favaro at edistar.com>
wrote:

> Thanks,
>
> I've tried with
>
>
>     <extension name="eavesdrop">
>         <condition field="destination_number"
> expression="^88(.*)$|^\*0(.*)$">
>             <action application="answer"/>
>             <action application="log" data="$1" />
>            <action application="userspy" data="user/$1"/>
>        </condition>
>     </extension>
>
> but unfortunately I have the same problem.
>
> SF.
>
> https://pastebin.freeswitch.org/24556
>
>
> ------------------------------
> *Da: *"?talo Rossi" <italo at freeswitch.org>
> *A: *"FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> *Inviato: *Luned?, 15 febbraio 2016 20:57:29
>
> *Oggetto: *Re: [Freeswitch-users] Problem with mod_spy
>
> It's probably due to the way you're dialing to your user, I can see in
> your log this bridge command: bridge(sofia/internal/4750%myserver)
>
> Just try to dial bridge user/4750 it'll work if 4750 is registered with FS.
>
> On Mon, Feb 15, 2016 at 6:12 AM, Stefano Favaro <
> stefano.favaro at edistar.com> wrote:
>
>> Thank you Italo.
>>
>> This is my log
>>
>> https://pastebin.freeswitch.org/24555
>>
>> and this is the id of my user spy call
>>
>> 94bae5a0-dd2c-4d0e-99fd-8ef81e753227
>>
>> Stefano
>>
>> ------------------------------
>> *Da: *"?talo Rossi" <italo at freeswitch.org>
>> *A: *"FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> *Inviato: *Gioved?, 11 febbraio 2016 18:52:46
>> *Oggetto: *Re: [Freeswitch-users] Problem with mod_spy
>>
>>
>> Stefano,
>>
>> Can you post your debug logs (/log 7)?
>>
>> Use https://pastebin.freeswitch.org/
>>
>> On Thu, Feb 11, 2016 at 1:33 PM, Stefano Favaro <
>> stefano.favaro at edistar.com> wrote:
>>
>>> Hello,
>>>
>>>  I have a problem with the mod_spy module.
>>> It seems that it just plays music and do not actually spy.
>>> I want to dial an extension with the number of the user I want to spy and wait for a call in or out for that user.
>>> Mod_spy application plays music but when a call is handled by the user I can't hear it continues to play the music on hold audio file.
>>>
>>> This is the dialplan:
>>>
>>> <extension name="eavesdrop">
>>>   <condition field="destination_number" expression="^88(.*)$|^\*0(.*)$">
>>>     <action application="answer"/>
>>>     <action application="userspy" data="$1 at myserver"/>
>>>   </condition>
>>> </extension>
>>>
>>>
>>> If I dial 881000, for example, It means I want to spy on user 1000.
>>> I have in and out calls from user 1000 but I can't hear.
>>> userspy_show in fs_cli, I get :
>>> 1000 at myserver : e9165eaf-e8d7-40ad-a8cd-d2963f363684
>>>
>>> 1 total spy
>>>
>>> I have FreeSwitch version FreeSWITCH Version 1.4.26-37~64bit (-37 64bit)
>>>
>>> SF.
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From royj at yandex.ru  Tue Feb 16 15:58:37 2016
From: royj at yandex.ru (royj at yandex.ru)
Date: Tue, 16 Feb 2016 15:58:37 +0300
Subject: [Freeswitch-users] hiredis: realm must be defined
Message-ID: <511661455627517@web2g.yandex.ru>

Hi, all
Can anybody clarify what mean 'hiredis: realm must be defined'? Where is the misconfiguration?
I am trying to use hiredis backend for limits and call api like 'api limit_usage hiredis myrealm id', got api/response '-1' and FreeSWITCH says in log 'hiredis: realm must be defined'.
Module mod_hiredis loaded and exists.
For hash backend works fine.


From aqsyounas at gmail.com  Tue Feb 16 16:35:12 2016
From: aqsyounas at gmail.com (Aqs Younas)
Date: Tue, 16 Feb 2016 18:35:12 +0500
Subject: [Freeswitch-users] 400 Bad To Header.
Message-ID: <CAGVOjVUgx9LuCO5Msyp7vLWUMq5T6JxB_mfGLE=g7dQCiHAnxw@mail.gmail.com>

Hi users,

My freeswitch is replying with me 400 Bad To Header whenever It receives
Bye containing this To Header.

To: 1111#15122031234 <sip:1111#15122031234 at 192.168.10.50:5060
>;tag=NKU2jrZvFaK0j

But I wonder in the case of Invite containing, this header everything is
fine.

Is there something wrong with this header? (#?)

Best regards.
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From mitch.capper at gmail.com  Tue Feb 16 17:22:44 2016
From: mitch.capper at gmail.com (Mitch Capper)
Date: Tue, 16 Feb 2016 06:22:44 -0800
Subject: [Freeswitch-users] freeswitch 1.6 on Ubuntu 14.04 LTS
In-Reply-To: <CAEJMVkAW1k18gkLv9JF6dpX6nkW5Bd0OiJM8Wab_BMcem4bqVw@mail.gmail.com>
References: <CAJsc=ovuAGfmj=TF4jyNEx7=f5sEYv36Wd1uTx=7mwfV=8VyEw@mail.gmail.com>
	<CALn233pSah+OPFr7MfdMbEk=skh3qvc9=-1giCq5C+bH4cD3QQ@mail.gmail.com>
	<CAEJMVkB1jrdgE5uj-FnCp6H963VTAhsBOvv0Gko_474C6oVeRQ@mail.gmail.com>
	<CAHCLHGsyMokNKjqv3D_XVj0Lte+a-Aoa5moX9mCzibNQHfRpXg@mail.gmail.com>
	<CAEJMVkAW1k18gkLv9JF6dpX6nkW5Bd0OiJM8Wab_BMcem4bqVw@mail.gmail.com>
Message-ID: <CALn233qVwdRN+=tPeqBcmZ5Hq2gFOYUkvv2HXNmTJ3r-LGPSQQ@mail.gmail.com>

Official packages sound great!
 For those who do want to compile from source for now a good starter may be
the dockerfile i use for ubuntu.   I simplified version is attached to the
ticket at:
https://freeswitch.org/jira/browse/FS-7833

you don't need to use docker the commands in the file(its just text), after
the "RUN " prefix are just shell commands so just copying and pasting them
into an ubuntu box should work.  I think that was from ubuntu 15 officially
but i don't think I changed the packages much from 14 (if you run into any
issues let me know and I can find my old one).

~mitch

On Mon, Feb 15, 2016 at 9:54 AM, Brian West <brian at freeswitch.org> wrote:

> I don't see why not, it may require more testing and some work....
>
> On Mon, Feb 15, 2016 at 11:39 AM, Tamas Jalsovszky <jalsot at gmail.com>
> wrote:
>
>> Are you going to support upcoming 16.04 LTS too?
>>
>> Jalsot
>>
>> On Mon, Feb 15, 2016 at 5:03 PM, Brian West <brian at freeswitch.org> wrote:
>>
>>> For those interested https://www.gofundme.com/freeswitch_ubuntu
>>>
>>> We are working on Official packages, if you care to donate to the cause.
>>>
>>> /b
>>>
>>> On Mon, Feb 15, 2016 at 9:51 AM, Mitch Capper <mitch.capper at gmail.com>
>>> wrote:
>>>
>>>> I haven't had an issue with 14 or 15 with FS with the stock modules
>>>> (and a few extras I use).
>>>>
>>>> ~mitch
>>>>
>>>> On Mon, Feb 15, 2016 at 6:34 AM, Caragea Silviu <silviu.cpp at gmail.com>
>>>> wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>> Is Ubuntu 14.04 supported on FS 1.6 ? Seems there are compilation
>>>>> issues for several modules
>>>>>
>>>>> 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
>>>>
>>>
>>>
>>>
>>> --
>>>
>>> *Brian West*
>>> brian at freeswitch.org
>>>
>>>
>>> *Twitter: @FreeSWITCH , @briankwest*
>>> http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>>
>>> 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
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> 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 stefano.favaro at edistar.com  Tue Feb 16 17:31:40 2016
From: stefano.favaro at edistar.com (Stefano Favaro)
Date: Tue, 16 Feb 2016 15:31:40 +0100 (CET)
Subject: [Freeswitch-users] Problem with mod_spy
In-Reply-To: <CABMX_w_SGWBLR9g3Y0skAKnkWprdM4kK3aypGGpfdk51DUHp3w@mail.gmail.com>
Message-ID: <361891616.7421.1455633100900.JavaMail.root@mailserver.edistar.com>

Ok, I see .... 

so, probably the problem is that I don't have any users already defined, they are runtime registered using xml_curl on an external server . 

Is there any other similar application that I can use to do the work? I mean conference or others? 

Thanks 




Stefano Favaro 
Sviluppo Servizi ed Applicazioni 
_____________________ 



Edistar Srl 


a socio unico soggetta a direzione e coordinamento di YourVoice SpA 


Via Artigianato 1 ? I ? 31050 Vedelago (TV) Italy 

Phone +39 0423 7331 ? Fax +39 0423 733133 

mobile: +39 3456636386 
skype: stefanofavaro 

www.edistar.com 
	

	


		


															



Le informazioni trasmesse attraverso la presente e-mail ed i suoi allegati sono dirette esclusivamente al destinatario e devono ritenersi riservate con divieto di diffusione e di uso nei giudizi salva espressa autorizzazione; nel caso di utilizzo senza espressa autorizzazione, potr? essere effettuata denuncia alla competente Autorit?. La diffusione e la comunicazione da parte di soggetto diverso dal destinatario ? vietata dall?art. 616 e ss. c.p. e dal d. l.vo n. 196/03. Se la presente e-mail e i suoi allegati fossero stati ricevuti per errore da persona diversa dal destinatario preghiamo di distruggere quanto ricevuto e di rinviare al mittente con lo stesso mezzo. Grazie per la collaborazione. 

This message may contain confidential and/or privileged information. If you are not the addressee or authorized to receive this for the addressee, you must not use, copy, disclose or take any action based on this message or any information herein. If you have received this message in error, please advise the sender immediately by reply e-mail and delete this message. Thank you for your cooperation. 

----- Messaggio originale -----
Da: "?talo Rossi" <italo at freeswitch.org> 
A: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org> 
Inviato: Marted?, 16 febbraio 2016 13:47:54 
Oggetto: Re: [Freeswitch-users] Problem with mod_spy 


I mean your bridge string, not userspy, instead of sofia/internal/4750 at ... just use bridge(user/4750) to DIAL to that user. Your userspy args looks good 


On Tue, Feb 16, 2016 at 5:40 AM, Stefano Favaro < stefano.favaro at edistar.com > wrote: 




Thanks, 

I've tried with 


<extension name="eavesdrop"> 
<condition field="destination_number" expression="^88(.*)$|^\*0(.*)$"> 
<action application="answer"/> 
<action application="log" data="$1" /> 
<action application="userspy" data="user/$1"/> 
</condition> 
</extension> 

but unfortunately I have the same problem. 

SF. 

https://pastebin.freeswitch.org/24556 



Da: "?talo Rossi" < italo at freeswitch.org > 
A: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org > 
Inviato: Luned?, 15 febbraio 2016 20:57:29 


Oggetto: Re: [Freeswitch-users] Problem with mod_spy 


It's probably due to the way you're dialing to your user, I can see in your log this bridge command: bridge(sofia/internal/4750%myserver) 


Just try to dial bridge user/4750 it'll work if 4750 is registered with FS. 


On Mon, Feb 15, 2016 at 6:12 AM, Stefano Favaro < stefano.favaro at edistar.com > wrote: 




Thank you Italo. 

This is my log 

https://pastebin.freeswitch.org/24555 

and this is the id of my user spy call 

94bae5a0-dd2c-4d0e-99fd-8ef81e753227 

Stefano 


Da: "?talo Rossi" < italo at freeswitch.org > 
A: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org > 
Inviato: Gioved?, 11 febbraio 2016 18:52:46 
Oggetto: Re: [Freeswitch-users] Problem with mod_spy 




Stefano, 


Can you post your debug logs (/log 7)? 


Use https://pastebin.freeswitch.org/ 


On Thu, Feb 11, 2016 at 1:33 PM, Stefano Favaro < stefano.favaro at edistar.com > wrote: 




Hello,

 I have a problem with the mod_spy module. 
It seems that it just plays music and do not actually spy.
I want to dial an extension with the number of the user I want to spy and wait for a call in or out for that user. 
Mod_spy application plays music but when a call is handled by the user I can't hear it continues to play the music on hold audio file. 

This is the dialplan:

<extension name="eavesdrop"> 
<condition field="destination_number" expression="^88(.*)$|^\*0(.*)$"> 
<action application="answer"/> 
<action application="userspy" data="$1 at myserver"/> 
</condition> 
</extension> 
If I dial 881000, for example, It means I want to spy on user 1000.
I have in and out calls from user 1000 but I can't hear. 
userspy_show in fs_cli, I get : 
1000 at myserver : e9165eaf-e8d7-40ad-a8cd-d2963f363684

1 total spy 

I have FreeSwitch version FreeSWITCH Version 1.4.26-37~64bit (-37 64bit) 

SF. 

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

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

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

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

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

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

FreeSWITCH-users mailing list 
FreeSWITCH-users at 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 veerabhadrarao.kankatala at panamaxil.com  Tue Feb 16 17:52:57 2016
From: veerabhadrarao.kankatala at panamaxil.com (Veerabhadrarao Kankatala)
Date: Tue, 16 Feb 2016 09:52:57 -0500 (EST)
Subject: [Freeswitch-users] WebRTC Core Dump for Video Conference.
Message-ID: <914027041.7946065.1455634377236.JavaMail.zimbra@panamaxil.com>

Hello, 

I am using freeswitch-1.6.5 version for webRTC 

I am using Chrome to intiate call (WebRTC Client) to another Video device (Grand stream), and i am using Default Dialplan to make call 

both are registered in freeswitch 

WebRTC Client: 1001 Default user 
GrandStream device: 1009 Default user 

When i make audio calling it is working fine, but when i make Video call from WebRTC Client ----> GrandStream device, my freeswitch is crashing 

following is the log 

Please tell me how to resolve this issue and let me know still anything i am missing. 


-- 
Thanks & Regards 
Veerabhadrarao Kankatala 
Software Developer (C & Unix ) 
PANAMAX INFOTECH LIMITED 
Mobile: +91-8401231249 
Messenger Id: Skype: veerabhadrarao.kankatala 
E-mail: veerabhadrarao.kankatala at panamaxil.com 

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From veerabhadrarao.kankatala at panamaxil.com  Tue Feb 16 17:54:43 2016
From: veerabhadrarao.kankatala at panamaxil.com (Veerabhadrarao Kankatala)
Date: Tue, 16 Feb 2016 09:54:43 -0500 (EST)
Subject: [Freeswitch-users] WebRTC Core Dump for Video Conference.
Message-ID: <1441201861.7946338.1455634483249.JavaMail.zimbra@panamaxil.com>

Hello, 

I am using freeswitch-1.6.5 version for webRTC 

I am using Chrome to intiate call (WebRTC Client) to another Video device (Grand stream), and i am using Default Dialplan to make call 

both are registered in freeswitch 

WebRTC Client: 1001 Default user 
GrandStream device: 1009 Default user 

When i make audio calling it is working fine, but when i make Video call from WebRTC Client ----> GrandStream device, my freeswitch is crashing 

Please find the attachment for Call Log to results the coreDump 

Please tell me how to resolve this issue and let me know still anything i am missing. 


-- 
Thanks & Regards 
Veerabhadrarao Kankatala 
Software Developer (C & Unix ) 
PANAMAX INFOTECH LIMITED 
Mobile: +91-8401231249 
Messenger Id: Skype: veerabhadrarao.kankatala 
E-mail: veerabhadrarao.kankatala at panamaxil.com 

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From krice at freeswitch.org  Tue Feb 16 18:00:40 2016
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 16 Feb 2016 09:00:40 -0600
Subject: [Freeswitch-users] WebRTC Core Dump for Video Conference.
In-Reply-To: <1441201861.7946338.1455634483249.JavaMail.zimbra@panamaxil.com>
References: <1441201861.7946338.1455634483249.JavaMail.zimbra@panamaxil.com>
Message-ID: <DFA26EC8-993A-4117-80D6-3428E7A8D989@freeswitch.org>

All crashes are bugs and all bugs go to jira

https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/5046304

Sent from my iPhone

> On Feb 16, 2016, at 8:54 AM, Veerabhadrarao Kankatala <veerabhadrarao.kankatala at panamaxil.com> wrote:
> 
> Hello,
> 
> I am using freeswitch-1.6.5 version for webRTC 
> 
> I am using Chrome to intiate call (WebRTC Client) to another Video device (Grand stream),  and i am using Default Dialplan to make call 
> 
> both are registered in freeswitch 
> 
> WebRTC Client:  1001 Default user
> GrandStream device: 1009 Default user
> 
> When i make audio calling it is working fine, but when i make Video call   from WebRTC Client ----> GrandStream device,  my freeswitch is crashing
> 
> Please find the attachment for Call Log to results the coreDump
> 
> Please tell me how to resolve this issue and let me know still anything i am missing.
> 
> 
> -- 
> Thanks & Regards
> Veerabhadrarao Kankatala
> Software Developer (C & Unix)
> PANAMAX INFOTECH LIMITED
> Mobile: +91-8401231249
> Messenger Id: Skype: veerabhadrarao.kankatala
> E-mail: veerabhadrarao.kankatala at panamaxil.com
> 
> <webRtc_Dump_log>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb 16 18:24:24 2016
From: mike at jerris.com (Michael Jerris)
Date: Tue, 16 Feb 2016 10:24:24 -0500
Subject: [Freeswitch-users] 400 Bad To Header.
In-Reply-To: <CAGVOjVUgx9LuCO5Msyp7vLWUMq5T6JxB_mfGLE=g7dQCiHAnxw@mail.gmail.com>
References: <CAGVOjVUgx9LuCO5Msyp7vLWUMq5T6JxB_mfGLE=g7dQCiHAnxw@mail.gmail.com>
Message-ID: <CAEwTsAiiSDrOYTMRoM=V0BZ_WCKmtEi_qRr0SqybAS+QDBKDAA@mail.gmail.com>

I thought the # had to be encoded.  Doesn't make sense that it wouldn't
give you the error every time

On Tuesday, February 16, 2016, Aqs Younas <aqsyounas at gmail.com> wrote:

> Hi users,
>
> My freeswitch is replying with me 400 Bad To Header whenever It receives
> Bye containing this To Header.
>
> To: 1111#15122031234 <sip:1111#15122031234 at 192.168.10.50:5060
> >;tag=NKU2jrZvFaK0j
>
> But I wonder in the case of Invite containing, this header everything is
> fine.
>
> Is there something wrong with this header? (#?)
>
> Best regards.
>
>
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From mike at jerris.com  Tue Feb 16 18:26:41 2016
From: mike at jerris.com (Michael Jerris)
Date: Tue, 16 Feb 2016 10:26:41 -0500
Subject: [Freeswitch-users] WebRTC Core Dump for Video Conference.
In-Reply-To: <DFA26EC8-993A-4117-80D6-3428E7A8D989@freeswitch.org>
References: <1441201861.7946338.1455634483249.JavaMail.zimbra@panamaxil.com>
	<DFA26EC8-993A-4117-80D6-3428E7A8D989@freeswitch.org>
Message-ID: <CAEwTsAi8aBUPkEMHUyo+YHQiOQZeNJao6BijxnZUrVdBKGwCPg@mail.gmail.com>

But not until you try the latest release and master to confirm we haven't
already fixed it

On Tuesday, February 16, 2016, Ken Rice <krice at freeswitch.org> wrote:

> All crashes are bugs and all bugs go to jira
>
>
> https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/5046304
>
> Sent from my iPhone
>
> On Feb 16, 2016, at 8:54 AM, Veerabhadrarao Kankatala <
> veerabhadrarao.kankatala at panamaxil.com
> <javascript:_e(%7B%7D,'cvml','veerabhadrarao.kankatala at panamaxil.com');>>
> wrote:
>
> Hello,
>
> I am using freeswitch-1.6.5 version for webRTC
>
> I am using Chrome to intiate call (WebRTC Client) to another Video device
> (Grand stream),  and i am using Default Dialplan to make call
>
> both are registered in freeswitch
>
> WebRTC Client:  1001 Default user
> GrandStream device: 1009 Default user
>
> When i make audio calling it is working fine, but when i make Video call
> from WebRTC Client ----> GrandStream device,  my freeswitch is crashing
>
> Please find the attachment for Call Log to results the coreDump
>
> Please tell me how to resolve this issue and let me know still anything i
> am missing.
>
>
> --
> Thanks & Regards
> *Veerabhadrarao Kankatala*
> Software Developer (C & Unix)
> *PANAMAX INFOTECH LIMITED*
> *Mobile:* *+91-8401231249*
> *Messenger Id:* Skype: veerabhadrarao.kankatala
> *E-mail: **veerabhadrarao.kankatala at panamaxil.com
> <javascript:_e(%7B%7D,'cvml','veerabhadrarao.kankatala at panamaxil.com');>*
>
> <webRtc_Dump_log>
>
> _________________________________________________________________________
> 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 tarantul at gmail.com  Tue Feb 16 14:52:44 2016
From: tarantul at gmail.com (Nick 'tarantul' Novikov)
Date: Tue, 16 Feb 2016 14:52:44 +0300
Subject: [Freeswitch-users] no audio after hold
Message-ID: <CAK+V7RxSdxaSLAE3iwUqSpQkTorED0X24PpG0YL3tEbOgah3bw@mail.gmail.com>

Hello

I have some problem with sound after hold usage. Sometime the sound
disappear after callcenter agent shift unhold.For callcenter agents we use
sip.js (version 0.7.2).
In freeswitch logs I see strange lines (
https://pastebin.freeswitch.org/24557).
Can anyone explain, what happened with audio and how to fix it?


-- 
tarantul
Dios es Amor
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From roman.kudinov at novelapp.com  Tue Feb 16 15:40:58 2016
From: roman.kudinov at novelapp.com (Roman Kudinov)
Date: Tue, 16 Feb 2016 15:40:58 +0300
Subject: [Freeswitch-users] problems with bridging a call,
	looks like transcoding is disabled
Message-ID: <56C318DA.7060603@novelapp.com>

Hi all,

I have a problem with bridging a call. My FS 1.6.6 is setup to bridge calls from RTMP-based source (using mod_rtmp) to a SIP gateway.
I have two branches in the dial plan.

1) One works through mod_conference which calls an outbound number using conference_set_auto_outcall

2) Another works by the direct bridging of incoming rtmp call into outbound SIP call.

Whilst the first branch works just fine, the second one does not. They both use the same sofia profiles, SIP gateways and outbound SIP numbers.
They both are called from the same RTMP source. Here are the snippet of codes.

================== This one works ====================================
> <extension name="conference_set_auto_outcall">
>    <condition field="destination_number" expression="123">
>      <action application="answer"/>
>      <action application="set" data="conference_auto_outcall_caller_id_name=$${effective_caller_id_name}"/>
>      <action application="set" data="conference_auto_outcall_caller_id_number=$${effective_caller_id_number}"/>
>      <action application="set" data="conference_auto_outcall_profile=default"/>
>      <action application="conference_set_auto_outcall" data="{ignore_early_media=true}sofia/gateway/sip_profile/number"/>
>      <action application="conference" data="$1+flags{moderator|endconf|mute}"/>
>    </condition>
> </extension>
===================================================

================ This one does not work ================
> <extension name="phone_only_session">
>    <condition field="destination_number" expression="456">
>      <action application="set" data="ignore_early_media=true"/>
>      <action application="set" data="absolute_codec_string=PCMU,PCMA,opus"/>
>      <action application="bridge" data="sofia/gateway/sip_profile/number"/>
>    </condition>
> </extension>
========================

I'd like to outline that they use the same SIP profiles, they are called from the same RTMP-source (they differs by the destination_number), they
call the same SIP number.
I turned on SIP tracing on and found that the call that is initiated by mod_conference offers the codecs according to outbound_codec_prefs set
in vars.xml, here is the piece of log:
>      m=audio 18684 RTP/AVP 0 102 103 104 105 8 101 106 108 110
>      a=rtpmap:0 PCMU/8000
>      a=rtpmap:102 SPEEX/8000
>      a=rtpmap:103 SPEEX/16000
>      a=rtpmap:104 SPEEX/32000
>      a=rtpmap:105 opus/48000/2
>      a=fmtp:105 useinbandfec=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40
>      a=rtpmap:8 PCMA/8000
>      a=rtpmap:101 telephone-event/8000
>      a=fmtp:101 0-16
>      a=rtpmap:106 telephone-event/16000
>      a=fmtp:106 0-16
>      a=rtpmap:108 telephone-event/32000
>      a=fmtp:108 0-16
>      a=rtpmap:110 telephone-event/48000
>      a=fmtp:110 0-16
>      a=ptime:20

But the directly bridged call offers incoming codec only, e.g. speex
>      m=audio 24972 RTP/AVP 102 101
>      a=rtpmap:102 SPEEX/16000
>      a=rtpmap:101 telephone-event/16000
>      a=fmtp:101 0-16
>      a=ptime:20

I tried everything I could imagine. I set absolute_codec_string in the dialplan (you can see it in the above snippet).
I explicitly set
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,G722,OPUS,PCMA"/>
in vars.xml

I tried to change
> <param name="inbound-codec-negotiation" value="generous"/>
from generous to greedy

I tried with true/false in the following parameters in internal.xml and
external.xml SOFIA profiles
> <param name="inbound-late-negotiation" value="false"/>
> <param name="inbound-zrtp-passthru" value="false"/>
Nothing changes.

Moreover the direct bridging worked fine on FS 1.4.7, it offered PCMU and Speex codecs to SIP.
I've upgraded to FS 1.6.6 and now it doesn't work. It looks like I missed an important setting that makes "bridge" application to work in
proxy media mode.
I checked I don't have neither bypass or proxy words in vars.xml, sofia.conf.xml, internal.xml, external.xml, public.xml or they are commented.

Does anybody have any ideas about the reason for such behavior?


Thanks,
Roman



From aqsyounas at gmail.com  Tue Feb 16 18:39:25 2016
From: aqsyounas at gmail.com (Aqs Younas)
Date: Tue, 16 Feb 2016 20:39:25 +0500
Subject: [Freeswitch-users] 400 Bad To Header.
In-Reply-To: <CAEwTsAiiSDrOYTMRoM=V0BZ_WCKmtEi_qRr0SqybAS+QDBKDAA@mail.gmail.com>
References: <CAGVOjVUgx9LuCO5Msyp7vLWUMq5T6JxB_mfGLE=g7dQCiHAnxw@mail.gmail.com>
	<CAEwTsAiiSDrOYTMRoM=V0BZ_WCKmtEi_qRr0SqybAS+QDBKDAA@mail.gmail.com>
Message-ID: <CAGVOjVVFrSLYvmB1a+Uy14cV9ybG-a9Y2a_oS+LFeU9wh2z5Vg@mail.gmail.com>

Thanks Michael,

What how can i encode this pound key? any hint is much  appreciated.

Best Regards

This email has been sent from a virus-free computer protected by Avast.
www.avast.com <https://www.avast.com/sig-email>
<#DDB4FAA8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

On 16 February 2016 at 20:24, Michael Jerris <mike at jerris.com> wrote:

> I thought the # had to be encoded.  Doesn't make sense that it wouldn't
> give you the error every time
>
>
> On Tuesday, February 16, 2016, Aqs Younas <aqsyounas at gmail.com> wrote:
>
>> Hi users,
>>
>> My freeswitch is replying with me 400 Bad To Header whenever It receives
>> Bye containing this To Header.
>>
>> To: 1111#15122031234 <sip:1111#15122031234 at 192.168.10.50:5060
>> >;tag=NKU2jrZvFaK0j
>>
>> But I wonder in the case of Invite containing, this header everything is
>> fine.
>>
>> Is there something wrong with this header? (#?)
>>
>> 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 brian at freeswitch.org  Tue Feb 16 19:00:46 2016
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Feb 2016 10:00:46 -0600
Subject: [Freeswitch-users] 400 Bad To Header.
In-Reply-To: <CAGVOjVVFrSLYvmB1a+Uy14cV9ybG-a9Y2a_oS+LFeU9wh2z5Vg@mail.gmail.com>
References: <CAGVOjVUgx9LuCO5Msyp7vLWUMq5T6JxB_mfGLE=g7dQCiHAnxw@mail.gmail.com>
	<CAEwTsAiiSDrOYTMRoM=V0BZ_WCKmtEi_qRr0SqybAS+QDBKDAA@mail.gmail.com>
	<CAGVOjVVFrSLYvmB1a+Uy14cV9ybG-a9Y2a_oS+LFeU9wh2z5Vg@mail.gmail.com>
Message-ID: <CAEJMVkAnayUun2c_kttf1ftsRE=Y8Z3gLDQXpRiuKiQLknXMEA@mail.gmail.com>

Its because its not quoted.

To: "1111#15122031234" <sip:1111#15122031234 at 192.168.10.50:5060
>;tag=NKU2jrZvFaK0j

would be correct, have your upstream FIX their broken stuff.

/b



On Tue, Feb 16, 2016 at 9:39 AM, Aqs Younas <aqsyounas at gmail.com> wrote:

> Thanks Michael,
>
> What how can i encode this pound key? any hint is much  appreciated.
>
> Best Regards
>
> This email has been sent from a virus-free computer protected by Avast.
> www.avast.com <https://www.avast.com/sig-email>
> <#-734131114_DDB4FAA8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>
> On 16 February 2016 at 20:24, Michael Jerris <mike at jerris.com> wrote:
>
>> I thought the # had to be encoded.  Doesn't make sense that it wouldn't
>> give you the error every time
>>
>>
>> On Tuesday, February 16, 2016, Aqs Younas <aqsyounas at gmail.com> wrote:
>>
>>> Hi users,
>>>
>>> My freeswitch is replying with me 400 Bad To Header whenever It receives
>>> Bye containing this To Header.
>>>
>>> To: 1111#15122031234 <sip:1111#15122031234 at 192.168.10.50:5060
>>> >;tag=NKU2jrZvFaK0j
>>>
>>> But I wonder in the case of Invite containing, this header everything is
>>> fine.
>>>
>>> Is there something wrong with this header? (#?)
>>>
>>> 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

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 mike at jerris.com  Tue Feb 16 19:02:02 2016
From: mike at jerris.com (Michael Jerris)
Date: Tue, 16 Feb 2016 11:02:02 -0500
Subject: [Freeswitch-users] 400 Bad To Header.
In-Reply-To: <CAGVOjVVFrSLYvmB1a+Uy14cV9ybG-a9Y2a_oS+LFeU9wh2z5Vg@mail.gmail.com>
References: <CAGVOjVUgx9LuCO5Msyp7vLWUMq5T6JxB_mfGLE=g7dQCiHAnxw@mail.gmail.com>
	<CAEwTsAiiSDrOYTMRoM=V0BZ_WCKmtEi_qRr0SqybAS+QDBKDAA@mail.gmail.com>
	<CAGVOjVVFrSLYvmB1a+Uy14cV9ybG-a9Y2a_oS+LFeU9wh2z5Vg@mail.gmail.com>
Message-ID: <FE1B7B9A-6455-4BBC-954A-44612866B98D@jerris.com>

You'll have to fix that on your device... 

> On Feb 16, 2016, at 10:39 AM, Aqs Younas <aqsyounas at gmail.com> wrote:
> 
> Thanks Michael, 
> 
> What how can i encode this pound key? any hint is much  appreciated. 
> 
> Best Regards
> 
> This email has been sent from a virus-free computer protected by Avast. 
> www.avast.com <https://www.avast.com/sig-email> <x-msg://10/#DDB4FAA8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
> 
> On 16 February 2016 at 20:24, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> I thought the # had to be encoded.  Doesn't make sense that it wouldn't give you the error every time 
> 
> 
> On Tuesday, February 16, 2016, Aqs Younas <aqsyounas at gmail.com <mailto:aqsyounas at gmail.com>> wrote:
> Hi users,
> 
> My freeswitch is replying with me 400 Bad To Header whenever It receives Bye containing this To Header. 
> 
> To: 1111#15122031234 <sip:1111#15122031234 at 192.168.10.50:5060 <http://15122031234 at 192.168.10.50:5060/>>;tag=NKU2jrZvFaK0j
> 
> But I wonder in the case of Invite containing, this header everything is fine. 
> 
> Is there something wrong with this header? (#?)
> 
> Best regards. 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From gmaruzz at gmail.com  Tue Feb 16 19:06:57 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 16 Feb 2016 17:06:57 +0100
Subject: [Freeswitch-users] no audio after hold
In-Reply-To: <CAK+V7RxSdxaSLAE3iwUqSpQkTorED0X24PpG0YL3tEbOgah3bw@mail.gmail.com>
References: <CAK+V7RxSdxaSLAE3iwUqSpQkTorED0X24PpG0YL3tEbOgah3bw@mail.gmail.com>
Message-ID: <CALXCt0rFM5xkK5JiwcqE5pj2pgFVkTaKZ8-qxcuqY4N1GGJLrA@mail.gmail.com>

Please pastebin a complete sip trace (from fs-cli: sofia global siptrace
on) and then put here the link to pastebin.
Il 16/Feb/2016 16:29, "Nick 'tarantul' Novikov" <tarantul at gmail.com> ha
scritto:

> Hello
>
> I have some problem with sound after hold usage. Sometime the sound
> disappear after callcenter agent shift unhold.For callcenter agents we use
> sip.js (version 0.7.2).
> In freeswitch logs I see strange lines (
> https://pastebin.freeswitch.org/24557).
> Can anyone explain, what happened with audio and how to fix it?
>
>
> --
> tarantul
> Dios es Amor
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb 16 19:07:26 2016
From: aqsyounas at gmail.com (Aqs Younas)
Date: Tue, 16 Feb 2016 21:07:26 +0500
Subject: [Freeswitch-users] 400 Bad To Header.
In-Reply-To: <FE1B7B9A-6455-4BBC-954A-44612866B98D@jerris.com>
References: <CAGVOjVUgx9LuCO5Msyp7vLWUMq5T6JxB_mfGLE=g7dQCiHAnxw@mail.gmail.com>
	<CAEwTsAiiSDrOYTMRoM=V0BZ_WCKmtEi_qRr0SqybAS+QDBKDAA@mail.gmail.com>
	<CAGVOjVVFrSLYvmB1a+Uy14cV9ybG-a9Y2a_oS+LFeU9wh2z5Vg@mail.gmail.com>
	<FE1B7B9A-6455-4BBC-954A-44612866B98D@jerris.com>
Message-ID: <CAGVOjVVVKs9rXXk9zVWsOzvHMHhwUX9USZ1_T-V+ZorJ+SiJKQ@mail.gmail.com>

Much Thanks :)
On 16-Feb-2016 9:04 pm, "Michael Jerris" <mike at jerris.com> wrote:

> You'll have to fix that on your device...
>
> On Feb 16, 2016, at 10:39 AM, Aqs Younas <aqsyounas at gmail.com> wrote:
>
> Thanks Michael,
>
> What how can i encode this pound key? any hint is much  appreciated.
>
> Best Regards
>
> This email has been sent from a virus-free computer protected by Avast.
> www.avast.com <https://www.avast.com/sig-email>
>
> On 16 February 2016 at 20:24, Michael Jerris <mike at jerris.com> wrote:
>
>> I thought the # had to be encoded.  Doesn't make sense that it wouldn't
>> give you the error every time
>>
>>
>> On Tuesday, February 16, 2016, Aqs Younas <aqsyounas at gmail.com> wrote:
>>
>>> Hi users,
>>>
>>> My freeswitch is replying with me 400 Bad To Header whenever It receives
>>> Bye containing this To Header.
>>>
>>> To: 1111#15122031234 <sip:1111#15122031234 at 192.168.10.50:5060
>>> >;tag=NKU2jrZvFaK0j
>>>
>>> But I wonder in the case of Invite containing, this header everything is
>>> fine.
>>>
>>> Is there something wrong with this header? (#?)
>>>
>>> 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  Tue Feb 16 19:11:47 2016
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Feb 2016 10:11:47 -0600
Subject: [Freeswitch-users] no audio after hold
In-Reply-To: <CALXCt0rFM5xkK5JiwcqE5pj2pgFVkTaKZ8-qxcuqY4N1GGJLrA@mail.gmail.com>
References: <CAK+V7RxSdxaSLAE3iwUqSpQkTorED0X24PpG0YL3tEbOgah3bw@mail.gmail.com>
	<CALXCt0rFM5xkK5JiwcqE5pj2pgFVkTaKZ8-qxcuqY4N1GGJLrA@mail.gmail.com>
Message-ID: <CAEJMVkCRrBU8DC=PFua2yb_N48-xBR3r1v9YSaCG5Ss7sV4=_w@mail.gmail.com>

Your logs are incomplete, need full debug logs.  Please don't filter them
excessively and try on master please.

On Tue, Feb 16, 2016 at 10:06 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> Please pastebin a complete sip trace (from fs-cli: sofia global siptrace
> on) and then put here the link to pastebin.
> Il 16/Feb/2016 16:29, "Nick 'tarantul' Novikov" <tarantul at gmail.com> ha
> scritto:
>
>> Hello
>>
>> I have some problem with sound after hold usage. Sometime the sound
>> disappear after callcenter agent shift unhold.For callcenter agents we use
>> sip.js (version 0.7.2).
>> In freeswitch logs I see strange lines (
>> https://pastebin.freeswitch.org/24557).
>> Can anyone explain, what happened with audio and how to fix it?
>>
>>
>> --
>> tarantul
>> Dios es Amor
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

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  Tue Feb 16 19:15:42 2016
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Feb 2016 10:15:42 -0600
Subject: [Freeswitch-users] problems with bridging a call,
 looks like transcoding is disabled
In-Reply-To: <56C318DA.7060603@novelapp.com>
References: <56C318DA.7060603@novelapp.com>
Message-ID: <CAEJMVkAe-HnbdUB9hJ9ZoR8xq2eKLqTwq_8PVJZhm_W0+TtoqA@mail.gmail.com>

This topic was talked about on the list in the past week:
https://freeswitch.org/jira/browse/FS-8321

"BEHAVIOR CHANGE Add variable media_mix_inbound_outbound_codecs to mix
inbound and outbound codecs"

/b

On Tue, Feb 16, 2016 at 6:40 AM, Roman Kudinov <roman.kudinov at novelapp.com>
wrote:

> Hi all,
>
> I have a problem with bridging a call. My FS 1.6.6 is setup to bridge
> calls from RTMP-based source (using mod_rtmp) to a SIP gateway.
> I have two branches in the dial plan.
>
> 1) One works through mod_conference which calls an outbound number using
> conference_set_auto_outcall
>
> 2) Another works by the direct bridging of incoming rtmp call into
> outbound SIP call.
>
> Whilst the first branch works just fine, the second one does not. They
> both use the same sofia profiles, SIP gateways and outbound SIP numbers.
> They both are called from the same RTMP source. Here are the snippet of
> codes.
>
> ================== This one works ====================================
> > <extension name="conference_set_auto_outcall">
> >    <condition field="destination_number" expression="123">
> >      <action application="answer"/>
> >      <action application="set"
> data="conference_auto_outcall_caller_id_name=$${effective_caller_id_name}"/>
> >      <action application="set"
> data="conference_auto_outcall_caller_id_number=$${effective_caller_id_number}"/>
> >      <action application="set"
> data="conference_auto_outcall_profile=default"/>
> >      <action application="conference_set_auto_outcall"
> data="{ignore_early_media=true}sofia/gateway/sip_profile/number"/>
> >      <action application="conference"
> data="$1+flags{moderator|endconf|mute}"/>
> >    </condition>
> > </extension>
> ===================================================
>
> ================ This one does not work ================
> > <extension name="phone_only_session">
> >    <condition field="destination_number" expression="456">
> >      <action application="set" data="ignore_early_media=true"/>
> >      <action application="set"
> data="absolute_codec_string=PCMU,PCMA,opus"/>
> >      <action application="bridge"
> data="sofia/gateway/sip_profile/number"/>
> >    </condition>
> > </extension>
> ========================
>
> I'd like to outline that they use the same SIP profiles, they are called
> from the same RTMP-source (they differs by the destination_number), they
> call the same SIP number.
> I turned on SIP tracing on and found that the call that is initiated by
> mod_conference offers the codecs according to outbound_codec_prefs set
> in vars.xml, here is the piece of log:
> >      m=audio 18684 RTP/AVP 0 102 103 104 105 8 101 106 108 110
> >      a=rtpmap:0 PCMU/8000
> >      a=rtpmap:102 SPEEX/8000
> >      a=rtpmap:103 SPEEX/16000
> >      a=rtpmap:104 SPEEX/32000
> >      a=rtpmap:105 opus/48000/2
> >      a=fmtp:105 useinbandfec=1; maxaveragebitrate=30000;
> maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40
> >      a=rtpmap:8 PCMA/8000
> >      a=rtpmap:101 telephone-event/8000
> >      a=fmtp:101 0-16
> >      a=rtpmap:106 telephone-event/16000
> >      a=fmtp:106 0-16
> >      a=rtpmap:108 telephone-event/32000
> >      a=fmtp:108 0-16
> >      a=rtpmap:110 telephone-event/48000
> >      a=fmtp:110 0-16
> >      a=ptime:20
>
> But the directly bridged call offers incoming codec only, e.g. speex
> >      m=audio 24972 RTP/AVP 102 101
> >      a=rtpmap:102 SPEEX/16000
> >      a=rtpmap:101 telephone-event/16000
> >      a=fmtp:101 0-16
> >      a=ptime:20
>
> I tried everything I could imagine. I set absolute_codec_string in the
> dialplan (you can see it in the above snippet).
> I explicitly set
> > <X-PRE-PROCESS cmd="set"
> data="outbound_codec_prefs=PCMU,G722,OPUS,PCMA"/>
> in vars.xml
>
> I tried to change
> > <param name="inbound-codec-negotiation" value="generous"/>
> from generous to greedy
>
> I tried with true/false in the following parameters in internal.xml and
> external.xml SOFIA profiles
> > <param name="inbound-late-negotiation" value="false"/>
> > <param name="inbound-zrtp-passthru" value="false"/>
> Nothing changes.
>
> Moreover the direct bridging worked fine on FS 1.4.7, it offered PCMU and
> Speex codecs to SIP.
> I've upgraded to FS 1.6.6 and now it doesn't work. It looks like I missed
> an important setting that makes "bridge" application to work in
> proxy media mode.
> I checked I don't have neither bypass or proxy words in vars.xml,
> sofia.conf.xml, internal.xml, external.xml, public.xml or they are
> commented.
>
> Does anybody have any ideas about the reason for such behavior?
>
>
> Thanks,
> Roman
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

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 decipher.hk at gmail.com  Tue Feb 16 19:40:29 2016
From: decipher.hk at gmail.com (=?utf-8?B?Um9kcmlnbyBSYW3DrXJleiBOb3JhbWJ1ZW5h?=)
Date: Tue, 16 Feb 2016 16:40:29 +0000
Subject: [Freeswitch-users] Answered/abandoned calls mod_callcenter
In-Reply-To: <27e84038904cd4a54ac63be19844adb2@mail2.boxtub.com>
References: <27e84038904cd4a54ac63be19844adb2@mail2.boxtub.com>
	<CABMX_w-2=si4EVhFvs+7OtickCj4eMaiu+zpPwgc4RMc4Snmyg@mail.gmail.com>
	<701581496dd4988b32ee3289cc2166df@mail2.boxtub.com>
Message-ID: <e706b42ad9df991db7106c09a58683c2@mail2.boxtub.com>

February 7 2016 3:18 PM, "Rodrigo Ram?rez Norambuena" <decipher.hk at gmail.com> wrote:
> February 5 2016 3:45 PM, "?talo Rossi" <italo at freeswitch.org> wrote:
> 
>> Hi Rodrigo,
> 
> Hi ?talo,
> 
>> We don't keep these counters in memory.
> 
> Ok.
> 
>> If you want to figure out right now how many calls were abandoned or answered you need to parse the
>> cdrs from these calls or listen to the ESL events and keep a counter.
> 
> I see for agents we have no_answer_count and calls_answered
> 
>> But, adding realtime counters to mod_callcenter shouldn't be difficult, a PR would be awesome ;)
> 
> I'll try.
> 
> What is the maximum line length columns?. In the [1] coding guidelines dont find
> 
> 1: https://freeswitch.org/confluence/display/FREESWITCH/Coding+Guidelines


Hi everyone,

I added support for QPanel of FreeSWITCH :)

You can test the branch fs into 
https://github.com/roramirez/qpanel/tree/fs

If you have any issue or improvement is welcome to hear.

Regards,


--
Rodrigo Ram?rez Norambuena
http://www.rodrigoramirez.com


From nneul at mst.edu  Tue Feb 16 20:20:06 2016
From: nneul at mst.edu (Nathan Neulinger)
Date: Tue, 16 Feb 2016 11:20:06 -0600
Subject: [Freeswitch-users] verto jsonrpc settings?
Message-ID: <56C35A46.2080509@mst.edu>


What does the verto/directory setting:

	<param name="jsonrpc-allowed-event-channels" value="demo,conference,presence"/>'

expose? It's very unclear from documentation.

Does this make ALL conference/presence events available to that user that logged in via verto or only those related to 
their particular registration/call?

-- Nathan

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


From roman.kudinov at novelapp.com  Tue Feb 16 20:53:41 2016
From: roman.kudinov at novelapp.com (Roman Kudinov)
Date: Tue, 16 Feb 2016 20:53:41 +0300
Subject: [Freeswitch-users] problems with bridging a call,
	looks like transcoding is disabled
In-Reply-To: <mailman.2239.1455639400.678.freeswitch-users@lists.freeswitch.org>
References: <mailman.2239.1455639400.678.freeswitch-users@lists.freeswitch.org>
Message-ID: <56C36225.4080906@novelapp.com>

Hi Brian,

I saw this variable but looks like used it the wrong way. Thanks for the 
help, the codecs negotiation works properly now!


16.02.2016 19:16, Brian West:
> This topic was talked about on the list in the past week: 
> https://freeswitch.org/jira/browse/FS-8321
>
> "BEHAVIOR CHANGE Add variable media_mix_inbound_outbound_codecs to mix 
> inbound and outbound codecs"
>
> /b
>
> On Tue, Feb 16, 2016 at 6:40 AM, Roman Kudinov 
> <roman.kudinov at novelapp.com <mailto:roman.kudinov at novelapp.com>> wrote:
>
>     Hi all,
>
>     I have a problem with bridging a call. My FS 1.6.6 is setup to
>     bridge calls from RTMP-based source (using mod_rtmp) to a SIP gateway.
>     I have two branches in the dial plan.
>
>     1) One works through mod_conference which calls an outbound number
>     using conference_set_auto_outcall
>
>     2) Another works by the direct bridging of incoming rtmp call into
>     outbound SIP call.
>
>     Whilst the first branch works just fine, the second one does not.
>     They both use the same sofia profiles, SIP gateways and outbound
>     SIP numbers.
>     They both are called from the same RTMP source. Here are the
>     snippet of codes.
>
>     ================== This one works ====================================
>     > <extension name="conference_set_auto_outcall">
>     >    <condition field="destination_number" expression="123">
>     >      <action application="answer"/>
>     >      <action application="set"
>     data="conference_auto_outcall_caller_id_name=$${effective_caller_id_name}"/>
>     >      <action application="set"
>     data="conference_auto_outcall_caller_id_number=$${effective_caller_id_number}"/>
>     >      <action application="set"
>     data="conference_auto_outcall_profile=default"/>
>     >      <action application="conference_set_auto_outcall"
>     data="{ignore_early_media=true}sofia/gateway/sip_profile/number"/>
>     >      <action application="conference"
>     data="$1+flags{moderator|endconf|mute}"/>
>     >    </condition>
>     > </extension>
>     ===================================================
>
>     ================ This one does not work ================
>     > <extension name="phone_only_session">
>     >    <condition field="destination_number" expression="456">
>     >      <action application="set" data="ignore_early_media=true"/>
>     >      <action application="set"
>     data="absolute_codec_string=PCMU,PCMA,opus"/>
>     >      <action application="bridge"
>     data="sofia/gateway/sip_profile/number"/>
>     >    </condition>
>     > </extension>
>     ========================
>
>     I'd like to outline that they use the same SIP profiles, they are
>     called from the same RTMP-source (they differs by the
>     destination_number), they
>     call the same SIP number.
>     I turned on SIP tracing on and found that the call that is
>     initiated by mod_conference offers the codecs according to
>     outbound_codec_prefs set
>     in vars.xml, here is the piece of log:
>     >      m=audio 18684 RTP/AVP 0 102 103 104 105 8 101 106 108 110
>     >      a=rtpmap:0 PCMU/8000
>     >      a=rtpmap:102 SPEEX/8000
>     >      a=rtpmap:103 SPEEX/16000
>     >      a=rtpmap:104 SPEEX/32000
>     >      a=rtpmap:105 opus/48000/2
>     >      a=fmtp:105 useinbandfec=1; maxaveragebitrate=30000;
>     maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40
>     >      a=rtpmap:8 PCMA/8000
>     >      a=rtpmap:101 telephone-event/8000
>     >      a=fmtp:101 0-16
>     >      a=rtpmap:106 telephone-event/16000
>     >      a=fmtp:106 0-16
>     >      a=rtpmap:108 telephone-event/32000
>     >      a=fmtp:108 0-16
>     >      a=rtpmap:110 telephone-event/48000
>     >      a=fmtp:110 0-16
>     >      a=ptime:20
>
>     But the directly bridged call offers incoming codec only, e.g. speex
>     >      m=audio 24972 RTP/AVP 102 101
>     >      a=rtpmap:102 SPEEX/16000
>     >      a=rtpmap:101 telephone-event/16000
>     >      a=fmtp:101 0-16
>     >      a=ptime:20
>
>     I tried everything I could imagine. I set absolute_codec_string in
>     the dialplan (you can see it in the above snippet).
>     I explicitly set
>     > <X-PRE-PROCESS cmd="set"
>     data="outbound_codec_prefs=PCMU,G722,OPUS,PCMA"/>
>     in vars.xml
>
>     I tried to change
>     > <param name="inbound-codec-negotiation" value="generous"/>
>     from generous to greedy
>
>     I tried with true/false in the following parameters in
>     internal.xml and
>     external.xml SOFIA profiles
>     > <param name="inbound-late-negotiation" value="false"/>
>     > <param name="inbound-zrtp-passthru" value="false"/>
>     Nothing changes.
>
>     Moreover the direct bridging worked fine on FS 1.4.7, it offered
>     PCMU and Speex codecs to SIP.
>     I've upgraded to FS 1.6.6 and now it doesn't work. It looks like I
>     missed an important setting that makes "bridge" application to work in
>     proxy media mode.
>     I checked I don't have neither bypass or proxy words in vars.xml,
>     sofia.conf.xml, internal.xml, external.xml, public.xml or they are
>     commented.
>
>     Does anybody have any ideas about the reason for such behavior?
>
>
>     Thanks,
>     Roman
>
>
>     _________________________________________________________________________
>     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
>
>
>
>
> -- 
>
> */Brian West/*
> brian at freeswitch.org <mailto:brian at freeswitch.org>
>
>
> */Twitter: @FreeSWITCH , @briankwest/*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> 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 blackc2004 at gmail.com  Tue Feb 16 22:14:58 2016
From: blackc2004 at gmail.com (Cj B)
Date: Tue, 16 Feb 2016 11:14:58 -0800
Subject: [Freeswitch-users] repeating: sofia_presence_sub_callback: endpt is
	internal
Message-ID: <54FDCCCA-1198-4513-A259-1D2ED84B68FA@gmail.com>

Hi everyone,

I just recently moved a setup from 1.4.26 to 1.6.6 and I keep seeing the following repeat over and over in the logs. 
2016-02-16 14:03:38.940698 [DEBUG] sofia_presence.c:2908 sofia_presence_sub_callback: endpt is internal
2016-02-16 14:03:38.940698 [DEBUG] sofia_presence.c:2908 sofia_presence_sub_callback: endpt is internal
http://pastebin.com/nxi9bH0N <http://pastebin.com/nxi9bH0N>
Calls appear to be completing properly and phones are registered. I was just wondering what this means and if there?s a way to stop it?

uname -a
Linux my.domainname.com <http://my.domainname.com/> 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt20-1+deb8u1 (2015-12-14) x86_64 GNU/Linux

/usr/local/freeswitch/bin/freeswitch -version
FreeSWITCH version: 1.6.6+git~20160111T201612Z~d2d0b3283a~64bit (git d2d0b32 2016-01-11 20:16:12Z 64bit)

Thanks,
Cj B

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From abaci64 at gmail.com  Tue Feb 16 22:15:27 2016
From: abaci64 at gmail.com (Abaci B)
Date: Tue, 16 Feb 2016 14:15:27 -0500
Subject: [Freeswitch-users] AMD
In-Reply-To: <CAEwTsAgXStGbsENve5fnYnhuVvfvGNbrVRv3fh1ieN1Q8MKwGw@mail.gmail.com>
References: <CAFmZFiNO_yzv8vwO9HFM41v-B7w7mHwzBBcJ74wGmZxEm+Nq7g@mail.gmail.com>
	<CAEwTsAgXStGbsENve5fnYnhuVvfvGNbrVRv3fh1ieN1Q8MKwGw@mail.gmail.com>
Message-ID: <CAFmZFiNrtBi8KcY5_peBjXq9oeBeGDagUyi4W-gdDsFiJyCAGw@mail.gmail.com>

I remember in the early days of FreeSWITCH it was recommended to turn off
hyperthreading as it had a negative impact on FreeSWITCH performace, Can
someone confirm if this has changed with newer processors or newer kernel,
if hyperthreading boosts performance then I would guess that a 10 core xeon
would give me at least as much performace as a 16 core AMD.
Thanks

On Sun, Feb 14, 2016 at 12:52 PM, Michael Jerris <mike at jerris.com> wrote:

> You would have to test to see, most of my experience is a couple chip
> generations back now but bang for the buck was much better with intel then
> and we saw many more weird issues on AMD boxes.
>
>
> On Sunday, February 14, 2016, Abaci B <abaci64 at gmail.com> wrote:
>
>> Hi all,
>> I know that in general Intel Xeon processors are recommended over AMD for
>> FreeSWITCH, my question is if AMD is really bad or just not as good as
>> Intel, I can get now a server with 4x AMD 6174 (48 Core total) for about
>> the same price I would pay for a Dual Xeon 6 Core (12 core total), Should I
>> go for the AMD which will give me 4x the amount of cores or stick to Xeon,
>> my goal is to get the maximum amount of channels/calls  (doing lua IVR with
>> no transcoding) and conferencing.
>> Any help, feedback, benchmarks or personal experience would be 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 gmaruzz at gmail.com  Tue Feb 16 22:25:52 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 16 Feb 2016 20:25:52 +0100
Subject: [Freeswitch-users] repeating: sofia_presence_sub_callback:
 endpt is internal
In-Reply-To: <54FDCCCA-1198-4513-A259-1D2ED84B68FA@gmail.com>
References: <54FDCCCA-1198-4513-A259-1D2ED84B68FA@gmail.com>
Message-ID: <CALXCt0qq=33b+afG-bAonLKkt01jsvMMGUgBU+Gx5GkH7YyMGg@mail.gmail.com>

Why you want to stop a debug message?
Il 16/Feb/2016 20:16, "Cj B" <blackc2004 at gmail.com> ha scritto:

> Hi everyone,
>
> I just recently moved a setup from 1.4.26 to 1.6.6 and I keep seeing the
> following repeat over and over in the logs.
>
>    1.
>
> 2016-02-16 14:03:38.940698 [DEBUG] sofia_presence.c:2908 sofia_presence_sub_callback: endpt is internal
> 2016-02-16 14:03:38.940698 [DEBUG] sofia_presence.c:2908 sofia_presence_sub_callback: endpt is internal
>
> http://pastebin.com/nxi9bH0N
>
> Calls appear to be completing properly and phones are registered. I was
> just wondering what this means and if there?s a way to stop it?
>
> uname -a
> Linux my.domainname.com 3.16.0-4-amd64 #1 SMP Debian
> 3.16.7-ckt20-1+deb8u1 (2015-12-14) x86_64 GNU/Linux
>
> /usr/local/freeswitch/bin/freeswitch -version
> FreeSWITCH version: 1.6.6+git~20160111T201612Z~d2d0b3283a~64bit (git
> d2d0b32 2016-01-11 20:16:12Z 64bit)
>
> Thanks,
> Cj B
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb 16 22:48:08 2016
From: mike at jerris.com (Michael Jerris)
Date: Tue, 16 Feb 2016 14:48:08 -0500
Subject: [Freeswitch-users] AMD
In-Reply-To: <CAFmZFiNrtBi8KcY5_peBjXq9oeBeGDagUyi4W-gdDsFiJyCAGw@mail.gmail.com>
References: <CAFmZFiNO_yzv8vwO9HFM41v-B7w7mHwzBBcJ74wGmZxEm+Nq7g@mail.gmail.com>
	<CAEwTsAgXStGbsENve5fnYnhuVvfvGNbrVRv3fh1ieN1Q8MKwGw@mail.gmail.com>
	<CAFmZFiNrtBi8KcY5_peBjXq9oeBeGDagUyi4W-gdDsFiJyCAGw@mail.gmail.com>
Message-ID: <C68581AC-1EF5-4EFA-ADCD-B7020313FCFF@jerris.com>

Have not tested it lately, but likely still true, in regards to "I guess"... why not just try it and see for sure?

> On Feb 16, 2016, at 2:15 PM, Abaci B <abaci64 at gmail.com> wrote:
> 
> I remember in the early days of FreeSWITCH it was recommended to turn off hyperthreading as it had a negative impact on FreeSWITCH performace, Can someone confirm if this has changed with newer processors or newer kernel, if hyperthreading boosts performance then I would guess that a 10 core xeon would give me at least as much performace as a 16 core AMD.
> Thanks
> 
> On Sun, Feb 14, 2016 at 12:52 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> You would have to test to see, most of my experience is a couple chip generations back now but bang for the buck was much better with intel then and we saw many more weird issues on AMD boxes.
> 
> 
> On Sunday, February 14, 2016, Abaci B <abaci64 at gmail.com <mailto:abaci64 at gmail.com>> wrote:
> Hi all,
> I know that in general Intel Xeon processors are recommended over AMD for FreeSWITCH, my question is if AMD is really bad or just not as good as Intel, I can get now a server with 4x AMD 6174 (48 Core total) for about the same price I would pay for a Dual Xeon 6 Core (12 core total), Should I go for the AMD which will give me 4x the amount of cores or stick to Xeon, my goal is to get the maximum amount of channels/calls  (doing lua IVR with no transcoding) and conferencing.
> Any help, feedback, benchmarks or personal experience would be appreciated
> Thanks

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From blackc2004 at gmail.com  Tue Feb 16 22:50:02 2016
From: blackc2004 at gmail.com (Cj B)
Date: Tue, 16 Feb 2016 11:50:02 -0800
Subject: [Freeswitch-users] repeating: sofia_presence_sub_callback:
	endpt is internal
In-Reply-To: <CALXCt0qq=33b+afG-bAonLKkt01jsvMMGUgBU+Gx5GkH7YyMGg@mail.gmail.com>
References: <54FDCCCA-1198-4513-A259-1D2ED84B68FA@gmail.com>
	<CALXCt0qq=33b+afG-bAonLKkt01jsvMMGUgBU+Gx5GkH7YyMGg@mail.gmail.com>
Message-ID: <DD3D7D80-EA1E-48A9-A4D8-3105EC1DFCD0@gmail.com>

I would like to know what it means so that if it?s a problem I can fix it.

Thanks
Cj B

> On Feb 16, 2016, at 11:25 AM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> 
> Why you want to stop a debug message?
> 
> Il 16/Feb/2016 20:16, "Cj B" <blackc2004 at gmail.com <mailto:blackc2004 at gmail.com>> ha scritto:
> Hi everyone,
> 
> I just recently moved a setup from 1.4.26 to 1.6.6 and I keep seeing the following repeat over and over in the logs. 
> 2016-02-16 14:03:38.940698 [DEBUG] sofia_presence.c:2908 sofia_presence_sub_callback: endpt is internal
> 2016-02-16 14:03:38.940698 [DEBUG] sofia_presence.c:2908 sofia_presence_sub_callback: endpt is internal
> http://pastebin.com/nxi9bH0N <http://pastebin.com/nxi9bH0N>
> Calls appear to be completing properly and phones are registered. I was just wondering what this means and if there?s a way to stop it?
> 
> uname -a
> Linux my.domainname.com <http://my.domainname.com/> 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt20-1+deb8u1 (2015-12-14) x86_64 GNU/Linux
> 
> /usr/local/freeswitch/bin/freeswitch -version
> FreeSWITCH version: 1.6.6+git~20160111T201612Z~d2d0b3283a~64bit (git d2d0b32 2016-01-11 20:16:12Z 64bit)
> 
> Thanks,
> Cj B
> 
> 
> _________________________________________________________________________
> 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  Tue Feb 16 22:55:23 2016
From: mike at jerris.com (Michael Jerris)
Date: Tue, 16 Feb 2016 14:55:23 -0500
Subject: [Freeswitch-users] repeating: sofia_presence_sub_callback:
	endpt is internal
In-Reply-To: <DD3D7D80-EA1E-48A9-A4D8-3105EC1DFCD0@gmail.com>
References: <54FDCCCA-1198-4513-A259-1D2ED84B68FA@gmail.com>
	<CALXCt0qq=33b+afG-bAonLKkt01jsvMMGUgBU+Gx5GkH7YyMGg@mail.gmail.com>
	<DD3D7D80-EA1E-48A9-A4D8-3105EC1DFCD0@gmail.com>
Message-ID: <7906696F-9366-4697-98BB-97EC561ED1F5@jerris.com>

From the block of code:

        if (is_dialog) {                                                                                                                                        
            // Usually we report the dialogs FROM the probed user.  The exception is when the monitored endpoint is internal,                                   
            // and its presence_id is set in the dialplan.  Reverse the direction if this is not a registered entity.                                           
            const char *caller = switch_str_nil(switch_event_get_header(helper->event, "caller-username"));                                                     
            if (!strcmp(direction, "inbound") && strcmp(sub_to_user,  caller)) {                                                                                
                // If inbound and the entity is not the caller (i.e. internal to FS), then the direction is reversed                                            
                // because it is not going through the B2BUA                                                                                                    
                switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "sofia_presence_sub_callback: endpt is internal\n");                                    
                direction = !strcasecmp(direction, "outbound") ? "inbound" : "outbound";                                                                        
            }                                                                                                                                                   
                                                                                                                                                                
        }                                                                                                                                                       

This is from a patch from 2011, and would have been in 1.4.26 as well, so you probably just didn't notice it before.  Are you actually having something not working?

> On Feb 16, 2016, at 2:50 PM, Cj B <blackc2004 at gmail.com> wrote:
> 
> I would like to know what it means so that if it?s a problem I can fix it.
> 
> Thanks
> Cj B
> 
>> On Feb 16, 2016, at 11:25 AM, Giovanni Maruzzelli <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>> wrote:
>> 
>> Why you want to stop a debug message?
>> 
>> Il 16/Feb/2016 20:16, "Cj B" <blackc2004 at gmail.com <mailto:blackc2004 at gmail.com>> ha scritto:
>> Hi everyone,
>> 
>> I just recently moved a setup from 1.4.26 to 1.6.6 and I keep seeing the following repeat over and over in the logs. 
>> 2016-02-16 14:03:38.940698 [DEBUG] sofia_presence.c:2908 sofia_presence_sub_callback: endpt is internal
>> 2016-02-16 14:03:38.940698 [DEBUG] sofia_presence.c:2908 sofia_presence_sub_callback: endpt is internal
>> http://pastebin.com/nxi9bH0N <http://pastebin.com/nxi9bH0N>
>> Calls appear to be completing properly and phones are registered. I was just wondering what this means and if there?s a way to stop it?
>> 
>> uname -a
>> Linux my.domainname.com <http://my.domainname.com/> 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt20-1+deb8u1 (2015-12-14) x86_64 GNU/Linux
>> 
>> /usr/local/freeswitch/bin/freeswitch -version
>> FreeSWITCH version: 1.6.6+git~20160111T201612Z~d2d0b3283a~64bit (git d2d0b32 2016-01-11 20:16:12Z 64bit)
>> 
>> Thanks,
>> Cj B
>> 
>> 
>> _________________________________________________________________________
>> 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
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From abaci64 at gmail.com  Tue Feb 16 23:01:35 2016
From: abaci64 at gmail.com (Abaci B)
Date: Tue, 16 Feb 2016 15:01:35 -0500
Subject: [Freeswitch-users] AMD
In-Reply-To: <C68581AC-1EF5-4EFA-ADCD-B7020313FCFF@jerris.com>
References: <CAFmZFiNO_yzv8vwO9HFM41v-B7w7mHwzBBcJ74wGmZxEm+Nq7g@mail.gmail.com>
	<CAEwTsAgXStGbsENve5fnYnhuVvfvGNbrVRv3fh1ieN1Q8MKwGw@mail.gmail.com>
	<CAFmZFiNrtBi8KcY5_peBjXq9oeBeGDagUyi4W-gdDsFiJyCAGw@mail.gmail.com>
	<C68581AC-1EF5-4EFA-ADCD-B7020313FCFF@jerris.com>
Message-ID: <CAFmZFiN4wpgugSumgq84eoq5kK2sxWx9uSpbNAgPf1N_OrvRqA@mail.gmail.com>

I'm debating between a 4x AMD 6276 (64 cores total) and 4x Intel E7-8867L
(40 cores/80 threads), I wish I would be able to afford both and see which
one (AMD vs Intel) is better... so if I know that hyperthreading is
boosting performance I would probably stick to intel as in general it's
better and it's more cores.

On Tue, Feb 16, 2016 at 2:48 PM, Michael Jerris <mike at jerris.com> wrote:

> Have not tested it lately, but likely still true, in regards to "I
> guess"... why not just try it and see for sure?
>
> On Feb 16, 2016, at 2:15 PM, Abaci B <abaci64 at gmail.com> wrote:
>
> I remember in the early days of FreeSWITCH it was recommended to turn off
> hyperthreading as it had a negative impact on FreeSWITCH performace, Can
> someone confirm if this has changed with newer processors or newer kernel,
> if hyperthreading boosts performance then I would guess that a 10 core xeon
> would give me at least as much performace as a 16 core AMD.
> Thanks
>
> On Sun, Feb 14, 2016 at 12:52 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> You would have to test to see, most of my experience is a couple chip
>> generations back now but bang for the buck was much better with intel then
>> and we saw many more weird issues on AMD boxes.
>>
>>
>> On Sunday, February 14, 2016, Abaci B <abaci64 at gmail.com> wrote:
>>
>>> Hi all,
>>> I know that in general Intel Xeon processors are recommended over AMD
>>> for FreeSWITCH, my question is if AMD is really bad or just not as good as
>>> Intel, I can get now a server with 4x AMD 6174 (48 Core total) for about
>>> the same price I would pay for a Dual Xeon 6 Core (12 core total), Should I
>>> go for the AMD which will give me 4x the amount of cores or stick to Xeon,
>>> my goal is to get the maximum amount of channels/calls  (doing lua IVR with
>>> no transcoding) and conferencing.
>>> Any help, feedback, benchmarks or personal experience would be
>>> 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 anthony.minessale at gmail.com  Tue Feb 16 23:07:36 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Feb 2016 14:07:36 -0600
Subject: [Freeswitch-users] AMD
In-Reply-To: <CAFmZFiN4wpgugSumgq84eoq5kK2sxWx9uSpbNAgPf1N_OrvRqA@mail.gmail.com>
References: <CAFmZFiNO_yzv8vwO9HFM41v-B7w7mHwzBBcJ74wGmZxEm+Nq7g@mail.gmail.com>
	<CAEwTsAgXStGbsENve5fnYnhuVvfvGNbrVRv3fh1ieN1Q8MKwGw@mail.gmail.com>
	<CAFmZFiNrtBi8KcY5_peBjXq9oeBeGDagUyi4W-gdDsFiJyCAGw@mail.gmail.com>
	<C68581AC-1EF5-4EFA-ADCD-B7020313FCFF@jerris.com>
	<CAFmZFiN4wpgugSumgq84eoq5kK2sxWx9uSpbNAgPf1N_OrvRqA@mail.gmail.com>
Message-ID: <CAKbxfG8AEZFaYGC9x6jYEp9uFUV3wfxpOLGJzAM34AJKtW2bXg@mail.gmail.com>

If you want an opinionated answer to help you decide:

Always use Intel.  AMD is Intel's R&D department.  The performance is
better and the hyperthreading on modern Intel is fine.


On Tue, Feb 16, 2016 at 2:01 PM, Abaci B <abaci64 at gmail.com> wrote:

> I'm debating between a 4x AMD 6276 (64 cores total) and 4x Intel E7-8867L
> (40 cores/80 threads), I wish I would be able to afford both and see which
> one (AMD vs Intel) is better... so if I know that hyperthreading is
> boosting performance I would probably stick to intel as in general it's
> better and it's more cores.
>
> On Tue, Feb 16, 2016 at 2:48 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> Have not tested it lately, but likely still true, in regards to "I
>> guess"... why not just try it and see for sure?
>>
>> On Feb 16, 2016, at 2:15 PM, Abaci B <abaci64 at gmail.com> wrote:
>>
>> I remember in the early days of FreeSWITCH it was recommended to turn off
>> hyperthreading as it had a negative impact on FreeSWITCH performace, Can
>> someone confirm if this has changed with newer processors or newer kernel,
>> if hyperthreading boosts performance then I would guess that a 10 core xeon
>> would give me at least as much performace as a 16 core AMD.
>> Thanks
>>
>> On Sun, Feb 14, 2016 at 12:52 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> You would have to test to see, most of my experience is a couple chip
>>> generations back now but bang for the buck was much better with intel then
>>> and we saw many more weird issues on AMD boxes.
>>>
>>>
>>> On Sunday, February 14, 2016, Abaci B <abaci64 at gmail.com> wrote:
>>>
>>>> Hi all,
>>>> I know that in general Intel Xeon processors are recommended over AMD
>>>> for FreeSWITCH, my question is if AMD is really bad or just not as good as
>>>> Intel, I can get now a server with 4x AMD 6174 (48 Core total) for about
>>>> the same price I would pay for a Dual Xeon 6 Core (12 core total), Should I
>>>> go for the AMD which will give me 4x the amount of cores or stick to Xeon,
>>>> my goal is to get the maximum amount of channels/calls  (doing lua IVR with
>>>> no transcoding) and conferencing.
>>>> Any help, feedback, benchmarks or personal experience would be
>>>> 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
>



-- 
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 blackc2004 at gmail.com  Tue Feb 16 23:07:32 2016
From: blackc2004 at gmail.com (Cj B)
Date: Tue, 16 Feb 2016 12:07:32 -0800
Subject: [Freeswitch-users] repeating: sofia_presence_sub_callback:
	endpt is internal
In-Reply-To: <7906696F-9366-4697-98BB-97EC561ED1F5@jerris.com>
References: <54FDCCCA-1198-4513-A259-1D2ED84B68FA@gmail.com>
	<CALXCt0qq=33b+afG-bAonLKkt01jsvMMGUgBU+Gx5GkH7YyMGg@mail.gmail.com>
	<DD3D7D80-EA1E-48A9-A4D8-3105EC1DFCD0@gmail.com>
	<7906696F-9366-4697-98BB-97EC561ED1F5@jerris.com>
Message-ID: <3706BD46-5D4D-4C3B-97E3-2821B5B7F7C3@gmail.com>

Thanks Michael. So far I don?t see anything not working but it?s only been a few hours since I turned on this server to start accepting real traffic.

But based on this code, it looks to me like it should only be happening during a call, correct? It looks to me like I?m getting this log repeating even when there is no call traffic on the server, like it?s happening on phone register.

Thanks
Cj B

> On Feb 16, 2016, at 11:55 AM, Michael Jerris <mike at jerris.com> wrote:
> 
> From the block of code:
> 
>         if (is_dialog) {                                                                                                                                        
>             // Usually we report the dialogs FROM the probed user.  The exception is when the monitored endpoint is internal,                                   
>             // and its presence_id is set in the dialplan.  Reverse the direction if this is not a registered entity.                                           
>             const char *caller = switch_str_nil(switch_event_get_header(helper->event, "caller-username"));                                                     
>             if (!strcmp(direction, "inbound") && strcmp(sub_to_user,  caller)) {                                                                                
>                 // If inbound and the entity is not the caller (i.e. internal to FS), then the direction is reversed                                            
>                 // because it is not going through the B2BUA                                                                                                    
>                 switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "sofia_presence_sub_callback: endpt is internal\n");                                    
>                 direction = !strcasecmp(direction, "outbound") ? "inbound" : "outbound";                                                                        
>             }                                                                                                                                                   
>                                                                                                                                                                 
>         }                                                                                                                                                       
> 
> This is from a patch from 2011, and would have been in 1.4.26 as well, so you probably just didn't notice it before.  Are you actually having something not working?
> 
>> On Feb 16, 2016, at 2:50 PM, Cj B <blackc2004 at gmail.com <mailto:blackc2004 at gmail.com>> wrote:
>> 
>> I would like to know what it means so that if it?s a problem I can fix it.
>> 
>> Thanks
>> Cj B
>> 
>>> On Feb 16, 2016, at 11:25 AM, Giovanni Maruzzelli <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>> wrote:
>>> 
>>> Why you want to stop a debug message?
>>> 
>>> Il 16/Feb/2016 20:16, "Cj B" <blackc2004 at gmail.com <mailto:blackc2004 at gmail.com>> ha scritto:
>>> Hi everyone,
>>> 
>>> I just recently moved a setup from 1.4.26 to 1.6.6 and I keep seeing the following repeat over and over in the logs. 
>>> 2016-02-16 14:03:38.940698 [DEBUG] sofia_presence.c:2908 sofia_presence_sub_callback: endpt is internal
>>> 2016-02-16 14:03:38.940698 [DEBUG] sofia_presence.c:2908 sofia_presence_sub_callback: endpt is internal
>>> http://pastebin.com/nxi9bH0N <http://pastebin.com/nxi9bH0N>
>>> Calls appear to be completing properly and phones are registered. I was just wondering what this means and if there?s a way to stop it?
>>> 
>>> uname -a
>>> Linux my.domainname.com <http://my.domainname.com/> 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt20-1+deb8u1 (2015-12-14) x86_64 GNU/Linux
>>> 
>>> /usr/local/freeswitch/bin/freeswitch -version
>>> FreeSWITCH version: 1.6.6+git~20160111T201612Z~d2d0b3283a~64bit (git d2d0b32 2016-01-11 20:16:12Z 64bit)
>>> 
>>> Thanks,
>>> Cj B
>>> 
>>> 
>>> _________________________________________________________________________
>>> 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://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/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
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From abaci64 at gmail.com  Tue Feb 16 23:13:51 2016
From: abaci64 at gmail.com (Abaci B)
Date: Tue, 16 Feb 2016 15:13:51 -0500
Subject: [Freeswitch-users] AMD
In-Reply-To: <CAKbxfG8AEZFaYGC9x6jYEp9uFUV3wfxpOLGJzAM34AJKtW2bXg@mail.gmail.com>
References: <CAFmZFiNO_yzv8vwO9HFM41v-B7w7mHwzBBcJ74wGmZxEm+Nq7g@mail.gmail.com>
	<CAEwTsAgXStGbsENve5fnYnhuVvfvGNbrVRv3fh1ieN1Q8MKwGw@mail.gmail.com>
	<CAFmZFiNrtBi8KcY5_peBjXq9oeBeGDagUyi4W-gdDsFiJyCAGw@mail.gmail.com>
	<C68581AC-1EF5-4EFA-ADCD-B7020313FCFF@jerris.com>
	<CAFmZFiN4wpgugSumgq84eoq5kK2sxWx9uSpbNAgPf1N_OrvRqA@mail.gmail.com>
	<CAKbxfG8AEZFaYGC9x6jYEp9uFUV3wfxpOLGJzAM34AJKtW2bXg@mail.gmail.com>
Message-ID: <CAFmZFiPv9WxsXvU2zO=FJqa581-mMJb95Evmg-RqhQRhrWHsxg@mail.gmail.com>

is the E7-8867L (Westmere / LGA1567 / Apr/3/11) modern/new enough to be
considered good for hyperthreading?

On Tue, Feb 16, 2016 at 3:07 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> If you want an opinionated answer to help you decide:
>
> Always use Intel.  AMD is Intel's R&D department.  The performance is
> better and the hyperthreading on modern Intel is fine.
>
>
> On Tue, Feb 16, 2016 at 2:01 PM, Abaci B <abaci64 at gmail.com> wrote:
>
>> I'm debating between a 4x AMD 6276 (64 cores total) and 4x Intel E7-8867L
>> (40 cores/80 threads), I wish I would be able to afford both and see which
>> one (AMD vs Intel) is better... so if I know that hyperthreading is
>> boosting performance I would probably stick to intel as in general it's
>> better and it's more cores.
>>
>> On Tue, Feb 16, 2016 at 2:48 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> Have not tested it lately, but likely still true, in regards to "I
>>> guess"... why not just try it and see for sure?
>>>
>>> On Feb 16, 2016, at 2:15 PM, Abaci B <abaci64 at gmail.com> wrote:
>>>
>>> I remember in the early days of FreeSWITCH it was recommended to turn
>>> off hyperthreading as it had a negative impact on FreeSWITCH performace,
>>> Can someone confirm if this has changed with newer processors or newer
>>> kernel, if hyperthreading boosts performance then I would guess that a 10
>>> core xeon would give me at least as much performace as a 16 core AMD.
>>> Thanks
>>>
>>> On Sun, Feb 14, 2016 at 12:52 PM, Michael Jerris <mike at jerris.com>
>>> wrote:
>>>
>>>> You would have to test to see, most of my experience is a couple chip
>>>> generations back now but bang for the buck was much better with intel then
>>>> and we saw many more weird issues on AMD boxes.
>>>>
>>>>
>>>> On Sunday, February 14, 2016, Abaci B <abaci64 at gmail.com> wrote:
>>>>
>>>>> Hi all,
>>>>> I know that in general Intel Xeon processors are recommended over AMD
>>>>> for FreeSWITCH, my question is if AMD is really bad or just not as good as
>>>>> Intel, I can get now a server with 4x AMD 6174 (48 Core total) for about
>>>>> the same price I would pay for a Dual Xeon 6 Core (12 core total), Should I
>>>>> go for the AMD which will give me 4x the amount of cores or stick to Xeon,
>>>>> my goal is to get the maximum amount of channels/calls  (doing lua IVR with
>>>>> no transcoding) and conferencing.
>>>>> Any help, feedback, benchmarks or personal experience would be
>>>>> 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
>>
>
>
>
> --
> 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 mike at jerris.com  Tue Feb 16 23:25:50 2016
From: mike at jerris.com (Michael Jerris)
Date: Tue, 16 Feb 2016 15:25:50 -0500
Subject: [Freeswitch-users] repeating: sofia_presence_sub_callback:
	endpt is internal
In-Reply-To: <3706BD46-5D4D-4C3B-97E3-2821B5B7F7C3@gmail.com>
References: <54FDCCCA-1198-4513-A259-1D2ED84B68FA@gmail.com>
	<CALXCt0qq=33b+afG-bAonLKkt01jsvMMGUgBU+Gx5GkH7YyMGg@mail.gmail.com>
	<DD3D7D80-EA1E-48A9-A4D8-3105EC1DFCD0@gmail.com>
	<7906696F-9366-4697-98BB-97EC561ED1F5@jerris.com>
	<3706BD46-5D4D-4C3B-97E3-2821B5B7F7C3@gmail.com>
Message-ID: <C9DC5A15-6BF7-4FAD-B8B3-8256D7295631@jerris.com>

its related to presence code...

> On Feb 16, 2016, at 3:07 PM, Cj B <blackc2004 at gmail.com> wrote:
> 
> Thanks Michael. So far I don?t see anything not working but it?s only been a few hours since I turned on this server to start accepting real traffic.
> 
> But based on this code, it looks to me like it should only be happening during a call, correct? It looks to me like I?m getting this log repeating even when there is no call traffic on the server, like it?s happening on phone register.
> 
> Thanks
> Cj B
> 
>> On Feb 16, 2016, at 11:55 AM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>> 
>> From the block of code:
>> 
>>         if (is_dialog) {                                                                                                                                        
>>             // Usually we report the dialogs FROM the probed user.  The exception is when the monitored endpoint is internal,                                   
>>             // and its presence_id is set in the dialplan.  Reverse the direction if this is not a registered entity.                                           
>>             const char *caller = switch_str_nil(switch_event_get_header(helper->event, "caller-username"));                                                     
>>             if (!strcmp(direction, "inbound") && strcmp(sub_to_user,  caller)) {                                                                                
>>                 // If inbound and the entity is not the caller (i.e. internal to FS), then the direction is reversed                                            
>>                 // because it is not going through the B2BUA                                                                                                    
>>                 switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "sofia_presence_sub_callback: endpt is internal\n");                                    
>>                 direction = !strcasecmp(direction, "outbound") ? "inbound" : "outbound";                                                                        
>>             }                                                                                                                                                   
>>                                                                                                                                                                 
>>         }                                                                                                                                                       
>> 
>> This is from a patch from 2011, and would have been in 1.4.26 as well, so you probably just didn't notice it before.  Are you actually having something not working?
>> 
>>> On Feb 16, 2016, at 2:50 PM, Cj B <blackc2004 at gmail.com <mailto:blackc2004 at gmail.com>> wrote:
>>> 
>>> I would like to know what it means so that if it?s a problem I can fix it.
>>> 
>>> Thanks
>>> Cj B
>>> 
>>>> On Feb 16, 2016, at 11:25 AM, Giovanni Maruzzelli <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>> wrote:
>>>> 
>>>> Why you want to stop a debug message?
>>>> 
>>>> Il 16/Feb/2016 20:16, "Cj B" <blackc2004 at gmail.com <mailto:blackc2004 at gmail.com>> ha scritto:
>>>> Hi everyone,
>>>> 
>>>> I just recently moved a setup from 1.4.26 to 1.6.6 and I keep seeing the following repeat over and over in the logs. 
>>>> 2016-02-16 14:03:38.940698 [DEBUG] sofia_presence.c:2908 sofia_presence_sub_callback: endpt is internal
>>>> 2016-02-16 14:03:38.940698 [DEBUG] sofia_presence.c:2908 sofia_presence_sub_callback: endpt is internal
>>>> http://pastebin.com/nxi9bH0N <http://pastebin.com/nxi9bH0N>
>>>> Calls appear to be completing properly and phones are registered. I was just wondering what this means and if there?s a way to stop it?
>>>> 
>>>> uname -a
>>>> Linux my.domainname.com <http://my.domainname.com/> 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt20-1+deb8u1 (2015-12-14) x86_64 GNU/Linux
>>>> 
>>>> /usr/local/freeswitch/bin/freeswitch -version
>>>> FreeSWITCH version: 1.6.6+git~20160111T201612Z~d2d0b3283a~64bit (git d2d0b32 2016-01-11 20:16:12Z 64bit)
>>>> 
>>>> Thanks,
>>>> Cj B
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> 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://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/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://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/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
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mgg at giagnocavo.net  Tue Feb 16 23:47:46 2016
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Tue, 16 Feb 2016 20:47:46 +0000
Subject: [Freeswitch-users] AMD
In-Reply-To: <CAFmZFiPv9WxsXvU2zO=FJqa581-mMJb95Evmg-RqhQRhrWHsxg@mail.gmail.com>
References: <CAFmZFiNO_yzv8vwO9HFM41v-B7w7mHwzBBcJ74wGmZxEm+Nq7g@mail.gmail.com>
	<CAEwTsAgXStGbsENve5fnYnhuVvfvGNbrVRv3fh1ieN1Q8MKwGw@mail.gmail.com>
	<CAFmZFiNrtBi8KcY5_peBjXq9oeBeGDagUyi4W-gdDsFiJyCAGw@mail.gmail.com>
	<C68581AC-1EF5-4EFA-ADCD-B7020313FCFF@jerris.com>
	<CAFmZFiN4wpgugSumgq84eoq5kK2sxWx9uSpbNAgPf1N_OrvRqA@mail.gmail.com>
	<CAKbxfG8AEZFaYGC9x6jYEp9uFUV3wfxpOLGJzAM34AJKtW2bXg@mail.gmail.com>
	<CAFmZFiPv9WxsXvU2zO=FJqa581-mMJb95Evmg-RqhQRhrWHsxg@mail.gmail.com>
Message-ID: <DM2PR07MB702CCF66181D3D05CD4328CD4AD0@DM2PR07MB702.namprd07.prod.outlook.com>

If you?re actually doing one huge conference so you need all calls into a single FS instance, ignore this message.

I might be a broken record on this topic, but seriously consider running multiple FreeSWITCH instances and setting affinity to specific sockets or even groups of cores. I?d be surprised to hear a quad-socket system is price/perf competitive with multiple smaller boxes.

At a minimum, you want to partition on NUMA nodes, otherwise every single memory access needs to jump through other processors. For instance, if CPU0 wants memory that?s connected to CPU3, it needs to request it via CPU1 then CPU3. This is slow (a single hop introduces a ~40% penalty). After that, anytime threads are writing to the same part of memory, CPUs have to take ownership from each other?s caches. This can be as expensive as going to main memory (thus being as slow as having no cache) and kill perf.

Presumably with enough media load, you?d want to look into how your NIC is delivering packets, and distribute that accordingly.

On Xeon 5500 series (a while ago, on FS 1.2) we didn?t notice any real impact of hyperthreading either way. But keeping FS partitioned on groups of 4 threads (2 cores) seemed to be the best performing option. The increase in stability was very marked (versus one FS process across 2x Xeons, 16 threads).

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Abaci B
Sent: Tuesday, 16 February, 2016 14:14
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] AMD

is the E7-8867L (Westmere / LGA1567 / Apr/3/11) modern/new enough to be considered good for hyperthreading?

On Tue, Feb 16, 2016 at 3:07 PM, Anthony Minessale <anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>> wrote:
If you want an opinionated answer to help you decide:

Always use Intel.  AMD is Intel's R&D department.  The performance is better and the hyperthreading on modern Intel is fine.


On Tue, Feb 16, 2016 at 2:01 PM, Abaci B <abaci64 at gmail.com<mailto:abaci64 at gmail.com>> wrote:
I'm debating between a 4x AMD 6276 (64 cores total) and 4x Intel E7-8867L (40 cores/80 threads), I wish I would be able to afford both and see which one (AMD vs Intel) is better... so if I know that hyperthreading is boosting performance I would probably stick to intel as in general it's better and it's more cores.

On Tue, Feb 16, 2016 at 2:48 PM, Michael Jerris <mike at jerris.com<mailto:mike at jerris.com>> wrote:
Have not tested it lately, but likely still true, in regards to "I guess"... why not just try it and see for sure?

On Feb 16, 2016, at 2:15 PM, Abaci B <abaci64 at gmail.com<mailto:abaci64 at gmail.com>> wrote:

I remember in the early days of FreeSWITCH it was recommended to turn off hyperthreading as it had a negative impact on FreeSWITCH performace, Can someone confirm if this has changed with newer processors or newer kernel, if hyperthreading boosts performance then I would guess that a 10 core xeon would give me at least as much performace as a 16 core AMD.
Thanks

On Sun, Feb 14, 2016 at 12:52 PM, Michael Jerris <mike at jerris.com<mailto:mike at jerris.com>> wrote:
You would have to test to see, most of my experience is a couple chip generations back now but bang for the buck was much better with intel then and we saw many more weird issues on AMD boxes.


On Sunday, February 14, 2016, Abaci B <abaci64 at gmail.com<mailto:abaci64 at gmail.com>> wrote:
Hi all,
I know that in general Intel Xeon processors are recommended over AMD for FreeSWITCH, my question is if AMD is really bad or just not as good as Intel, I can get now a server with 4x AMD 6174 (48 Core total) for about the same price I would pay for a Dual Xeon 6 Core (12 core total), Should I go for the AMD which will give me 4x the amount of cores or stick to Xeon, my goal is to get the maximum amount of channels/calls  (doing lua IVR with no transcoding) and conferencing.
Any help, feedback, benchmarks or personal experience would be appreciated
Thanks


_________________________________________________________________________
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



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

? http://freeswitch.org/  ? http://cluecon.com/  ? http://twitter.com/FreeSWITCH
? irc.freenode.net<http://irc.freenode.net> #freeswitch ? http://freeswitch.org/g+
ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>  ? +19193869900<tel:%2B19193869900>

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA

_________________________________________________________________________
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
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http://www.freeswitch.org

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From brian at freeswitch.org  Wed Feb 17 00:00:02 2016
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Feb 2016 15:00:02 -0600
Subject: [Freeswitch-users] AMD
In-Reply-To: <C68581AC-1EF5-4EFA-ADCD-B7020313FCFF@jerris.com>
References: <CAFmZFiNO_yzv8vwO9HFM41v-B7w7mHwzBBcJ74wGmZxEm+Nq7g@mail.gmail.com>
	<CAEwTsAgXStGbsENve5fnYnhuVvfvGNbrVRv3fh1ieN1Q8MKwGw@mail.gmail.com>
	<CAFmZFiNrtBi8KcY5_peBjXq9oeBeGDagUyi4W-gdDsFiJyCAGw@mail.gmail.com>
	<C68581AC-1EF5-4EFA-ADCD-B7020313FCFF@jerris.com>
Message-ID: <CAEJMVkCe5axzLx2zt5pGY5wVT3JFsE2EKsPWaQ6YdXTqPD5O-w@mail.gmail.com>

Early hyper-threading would have been an issue, but newer CPUs seem to do a
great job of it.

On Tue, Feb 16, 2016 at 1:48 PM, Michael Jerris <mike at jerris.com> wrote:

> Have not tested it lately, but likely still true, in regards to "I
> guess"... why not just try it and see for sure?
>
> On Feb 16, 2016, at 2:15 PM, Abaci B <abaci64 at gmail.com> wrote:
>
> I remember in the early days of FreeSWITCH it was recommended to turn off
> hyperthreading as it had a negative impact on FreeSWITCH performace, Can
> someone confirm if this has changed with newer processors or newer kernel,
> if hyperthreading boosts performance then I would guess that a 10 core xeon
> would give me at least as much performace as a 16 core AMD.
> Thanks
>
> On Sun, Feb 14, 2016 at 12:52 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> You would have to test to see, most of my experience is a couple chip
>> generations back now but bang for the buck was much better with intel then
>> and we saw many more weird issues on AMD boxes.
>>
>>
>> On Sunday, February 14, 2016, Abaci B <abaci64 at gmail.com> wrote:
>>
>>> Hi all,
>>> I know that in general Intel Xeon processors are recommended over AMD
>>> for FreeSWITCH, my question is if AMD is really bad or just not as good as
>>> Intel, I can get now a server with 4x AMD 6174 (48 Core total) for about
>>> the same price I would pay for a Dual Xeon 6 Core (12 core total), Should I
>>> go for the AMD which will give me 4x the amount of cores or stick to Xeon,
>>> my goal is to get the maximum amount of channels/calls  (doing lua IVR with
>>> no transcoding) and conferencing.
>>> Any help, feedback, benchmarks or personal experience would be
>>> 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
>



-- 

*Brian West*
brian at freeswitch.org


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

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 abaci64 at gmail.com  Wed Feb 17 00:03:59 2016
From: abaci64 at gmail.com (Abaci B)
Date: Tue, 16 Feb 2016 16:03:59 -0500
Subject: [Freeswitch-users] AMD
In-Reply-To: <DM2PR07MB702CCF66181D3D05CD4328CD4AD0@DM2PR07MB702.namprd07.prod.outlook.com>
References: <CAFmZFiNO_yzv8vwO9HFM41v-B7w7mHwzBBcJ74wGmZxEm+Nq7g@mail.gmail.com>
	<CAEwTsAgXStGbsENve5fnYnhuVvfvGNbrVRv3fh1ieN1Q8MKwGw@mail.gmail.com>
	<CAFmZFiNrtBi8KcY5_peBjXq9oeBeGDagUyi4W-gdDsFiJyCAGw@mail.gmail.com>
	<C68581AC-1EF5-4EFA-ADCD-B7020313FCFF@jerris.com>
	<CAFmZFiN4wpgugSumgq84eoq5kK2sxWx9uSpbNAgPf1N_OrvRqA@mail.gmail.com>
	<CAKbxfG8AEZFaYGC9x6jYEp9uFUV3wfxpOLGJzAM34AJKtW2bXg@mail.gmail.com>
	<CAFmZFiPv9WxsXvU2zO=FJqa581-mMJb95Evmg-RqhQRhrWHsxg@mail.gmail.com>
	<DM2PR07MB702CCF66181D3D05CD4328CD4AD0@DM2PR07MB702.namprd07.prod.outlook.com>
Message-ID: <CAFmZFiNObsZKSbSisHyPjCRcVdabO6pgM8yyL7wRscx7W1=PDw@mail.gmail.com>

Running one huge conference is actually part of the reason that I want to
go with a huge box, but your input is actually very informative as I do
intend to use multiple instance of freeswitch (using proxmox / lxc) yet
still have the ability on occasion to allocate lots of CPU to a single
instance.
You're right that multiple smaller machines will probably win in
price/performance, colo space is also a factor.
Thanks Anthony and MIchael for all the info. once I have server up and
running I will try to test the limits and report back.

On Tue, Feb 16, 2016 at 3:47 PM, Michael Giagnocavo <mgg at giagnocavo.net>
wrote:

> If you?re actually doing one huge conference so you need all calls into a
> single FS instance, ignore this message.
>
>
>
> I might be a broken record on this topic, but seriously consider running
> multiple FreeSWITCH instances and setting affinity to specific sockets or
> even groups of cores. I?d be surprised to hear a quad-socket system is
> price/perf competitive with multiple smaller boxes.
>
>
>
> At a minimum, you want to partition on NUMA nodes, otherwise every single
> memory access needs to jump through other processors. For instance, if CPU0
> wants memory that?s connected to CPU3, it needs to request it via CPU1 then
> CPU3. This is slow (a single hop introduces a ~40% penalty). After that,
> anytime threads are writing to the same part of memory, CPUs have to take
> ownership from each other?s caches. This can be as expensive as going to
> main memory (thus being as slow as having no cache) and kill perf.
>
>
>
> Presumably with enough media load, you?d want to look into how your NIC is
> delivering packets, and distribute that accordingly.
>
>
>
> On Xeon 5500 series (a while ago, on FS 1.2) we didn?t notice any real
> impact of hyperthreading either way. But keeping FS partitioned on groups
> of 4 threads (2 cores) seemed to be the best performing option. The
> increase in stability was very marked (versus one FS process across 2x
> Xeons, 16 threads).
>
>
>
> -Michael
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Abaci B
> *Sent:* Tuesday, 16 February, 2016 14:14
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] AMD
>
>
>
> is the E7-8867L (Westmere / LGA1567 / Apr/3/11) modern/new enough to be
> considered good for hyperthreading?
>
>
>
> On Tue, Feb 16, 2016 at 3:07 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
> If you want an opinionated answer to help you decide:
>
>
>
> Always use Intel.  AMD is Intel's R&D department.  The performance is
> better and the hyperthreading on modern Intel is fine.
>
>
>
>
>
> On Tue, Feb 16, 2016 at 2:01 PM, Abaci B <abaci64 at gmail.com> wrote:
>
> I'm debating between a 4x AMD 6276 (64 cores total) and 4x Intel E7-8867L
> (40 cores/80 threads), I wish I would be able to afford both and see which
> one (AMD vs Intel) is better... so if I know that hyperthreading is
> boosting performance I would probably stick to intel as in general it's
> better and it's more cores.
>
>
>
> On Tue, Feb 16, 2016 at 2:48 PM, Michael Jerris <mike at jerris.com> wrote:
>
> Have not tested it lately, but likely still true, in regards to "I
> guess"... why not just try it and see for sure?
>
>
>
> On Feb 16, 2016, at 2:15 PM, Abaci B <abaci64 at gmail.com> wrote:
>
>
>
> I remember in the early days of FreeSWITCH it was recommended to turn off
> hyperthreading as it had a negative impact on FreeSWITCH performace, Can
> someone confirm if this has changed with newer processors or newer kernel,
> if hyperthreading boosts performance then I would guess that a 10 core xeon
> would give me at least as much performace as a 16 core AMD.
>
> Thanks
>
>
>
> On Sun, Feb 14, 2016 at 12:52 PM, Michael Jerris <mike at jerris.com> wrote:
>
> You would have to test to see, most of my experience is a couple chip
> generations back now but bang for the buck was much better with intel then
> and we saw many more weird issues on AMD boxes.
>
>
>
> On Sunday, February 14, 2016, Abaci B <abaci64 at gmail.com> wrote:
>
> Hi all,
>
> I know that in general Intel Xeon processors are recommended over AMD for
> FreeSWITCH, my question is if AMD is really bad or just not as good as
> Intel, I can get now a server with 4x AMD 6174 (48 Core total) for about
> the same price I would pay for a Dual Xeon 6 Core (12 core total), Should I
> go for the AMD which will give me 4x the amount of cores or stick to Xeon,
> my goal is to get the maximum amount of channels/calls  (doing lua IVR with
> no transcoding) and conferencing.
>
> Any help, feedback, benchmarks or personal experience would be 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
>
>
>
>
>
> --
>
> 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
>
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From naveen.khanna.bm at gmail.com  Wed Feb 17 08:33:48 2016
From: naveen.khanna.bm at gmail.com (Naveen Khanna)
Date: Wed, 17 Feb 2016 11:03:48 +0530
Subject: [Freeswitch-users] BLF of dialing party
Message-ID: <B5D22562-8971-4EA4-BC90-A5263B3A523C@gmail.com>

HI,

I have come across a situation where BLF of dialing party is not lit. However, if the same user is receiving the Calls then we get proper BLF.


Regards,


Naveen Khanna
M : +91-9911393060 | Skype : naveen.khanna.bm





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From jungleboogie0 at gmail.com  Wed Feb 17 09:18:19 2016
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Tue, 16 Feb 2016 22:18:19 -0800
Subject: [Freeswitch-users] how to sip uri call
Message-ID: <CAKE2PDstUz9q+raO1f0dQgtR9JbKaU9gxGOhAvbQW6a5sznDfA@mail.gmail.com>

Hello All,

I'd like my freeswitch server to make sip uri calls to various URI's.

I see this in dialplan/default.xml:
   <!-- If sip_req_host is not a local domain then this has to be an
external sip uri -->
    <!--
    <extension name="external_sip_uri" continue="true">
      <condition field="source" expression="mod_sofia"/>
      <condition field="${outside_call}" expression="^$"/>
      <condition field="${domain_exists(${sip_req_host})}" expression="true">
        <anti-action application="bridge"
data="sofia/${use_profile}/${sip_to_uri}"/>
      </condition>
    </extension>
    -->

If I enable this, I can call any sip URI but then I can't call an
extension local to the FS instance nor out on the PSTN. I see in

In the same file, I do see this:
<action application="bridge"
  data="sofia/${use_profile}/$1 at conference.freeswitch.org"/>

But I think there must be a method to enable URI calling for any
domain--not just explicitly listed URI.


If there's a confluence page that already covers this, feel free to
send the article to me so I can try it out.


Thanks for any assistance!

-- 
-------
inum: 883510009027723
sip: jungleboogie at sip2sip.info
xmpp: jungle-boogie at jit.si


From ashwinrath at gmail.com  Wed Feb 17 09:20:43 2016
From: ashwinrath at gmail.com (Ashwin Rath)
Date: Wed, 17 Feb 2016 11:50:43 +0530
Subject: [Freeswitch-users] How to convert a json string returned from a
 webservice to a json object in lua
Message-ID: <CAA_Nwu4cW4DqJqc4RFvXdg32thu07b2rWpd=D_78nPqyyubCvw@mail.gmail.com>

I am using LUA to POST to a webservice and i receive a json string . How do
i convert this string to a json object ?

-- 
Ashwin Kumar Rath
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From krice at freeswitch.org  Wed Feb 17 09:26:12 2016
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 17 Feb 2016 00:26:12 -0600
Subject: [Freeswitch-users] how to sip uri call
In-Reply-To: <CAKE2PDstUz9q+raO1f0dQgtR9JbKaU9gxGOhAvbQW6a5sznDfA@mail.gmail.com>
References: <CAKE2PDstUz9q+raO1f0dQgtR9JbKaU9gxGOhAvbQW6a5sznDfA@mail.gmail.com>
Message-ID: <167301d1694c$194b20c0$4be16240$@freeswitch.org>

Its all about ordering of the dialplan... first thing that matches wins...



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of jungle
Boogie
Sent: Wednesday, February 17, 2016 12:18 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] how to sip uri call

Hello All,

I'd like my freeswitch server to make sip uri calls to various URI's.

I see this in dialplan/default.xml:
   <!-- If sip_req_host is not a local domain then this has to be an
external sip uri -->
    <!--
    <extension name="external_sip_uri" continue="true">
      <condition field="source" expression="mod_sofia"/>
      <condition field="${outside_call}" expression="^$"/>
      <condition field="${domain_exists(${sip_req_host})}"
expression="true">
        <anti-action application="bridge"
data="sofia/${use_profile}/${sip_to_uri}"/>
      </condition>
    </extension>
    -->

If I enable this, I can call any sip URI but then I can't call an extension
local to the FS instance nor out on the PSTN. I see in

In the same file, I do see this:
<action application="bridge"
  data="sofia/${use_profile}/$1 at conference.freeswitch.org"/>

But I think there must be a method to enable URI calling for any domain--not
just explicitly listed URI.


If there's a confluence page that already covers this, feel free to send the
article to me so I can try it out.


Thanks for any assistance!

--
-------
inum: 883510009027723
sip: jungleboogie at sip2sip.info
xmpp: jungle-boogie at jit.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



From tarantul at gmail.com  Wed Feb 17 13:03:18 2016
From: tarantul at gmail.com (Nick 'tarantul' Novikov)
Date: Wed, 17 Feb 2016 13:03:18 +0300
Subject: [Freeswitch-users] no audio after hold
In-Reply-To: <CAEJMVkCRrBU8DC=PFua2yb_N48-xBR3r1v9YSaCG5Ss7sV4=_w@mail.gmail.com>
References: <CAK+V7RxSdxaSLAE3iwUqSpQkTorED0X24PpG0YL3tEbOgah3bw@mail.gmail.com>
	<CALXCt0rFM5xkK5JiwcqE5pj2pgFVkTaKZ8-qxcuqY4N1GGJLrA@mail.gmail.com>
	<CAEJMVkCRrBU8DC=PFua2yb_N48-xBR3r1v9YSaCG5Ss7sV4=_w@mail.gmail.com>
Message-ID: <CAK+V7RygPe5Bi-xzabjSdgovbvwo++k=X+FwZ0nqNQaeGkKYFQ@mail.gmail.com>

Hello!

Log file is large, I can't paste it in FS pastebin.
You may download gzip log file here
https://yadi.sk/d/pLbEHi_LoyWsS

On Tue, Feb 16, 2016 at 7:11 PM, Brian West <brian at freeswitch.org> wrote:

> Your logs are incomplete, need full debug logs.  Please don't filter them
> excessively and try on master please.
>
> On Tue, Feb 16, 2016 at 10:06 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> Please pastebin a complete sip trace (from fs-cli: sofia global siptrace
>> on) and then put here the link to pastebin.
>> Il 16/Feb/2016 16:29, "Nick 'tarantul' Novikov" <tarantul at gmail.com> ha
>> scritto:
>>
>>> Hello
>>>
>>> I have some problem with sound after hold usage. Sometime the sound
>>> disappear after callcenter agent shift unhold.For callcenter agents we use
>>> sip.js (version 0.7.2).
>>> In freeswitch logs I see strange lines (
>>> https://pastebin.freeswitch.org/24557).
>>> Can anyone explain, what happened with audio and how to fix it?
>>>
>>>
>>> --
>>> tarantul
>>> Dios es Amor
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> 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
>



-- 
tarantul
Dios es Amor
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From royj at yandex.ru  Wed Feb 17 14:07:16 2016
From: royj at yandex.ru (royj at yandex.ru)
Date: Wed, 17 Feb 2016 14:07:16 +0300
Subject: [Freeswitch-users] early media after bridge
Message-ID: <28121455707236@web17o.yandex.ru>

Hi, all
Is there any chance to use the early media from the b-leg as ringback to the a-leg when a-leg is answered for play_and_get_digits for example?


From cmrienzo at gmail.com  Wed Feb 17 16:09:41 2016
From: cmrienzo at gmail.com (cmrienzo at gmail.com)
Date: Wed, 17 Feb 2016 08:09:41 -0500
Subject: [Freeswitch-users] How to convert a json string returned from a
	webservice to a json object in lua
In-Reply-To: <CAA_Nwu4cW4DqJqc4RFvXdg32thu07b2rWpd=D_78nPqyyubCvw@mail.gmail.com>
References: <CAA_Nwu4cW4DqJqc4RFvXdg32thu07b2rWpd=D_78nPqyyubCvw@mail.gmail.com>
Message-ID: <7837F9BB-9117-4AFA-AFBE-29D5A79A26AD@gmail.com>

dkjson w/ lpeg works well for me. 

> On Feb 17, 2016, at 01:20, Ashwin Rath <ashwinrath at gmail.com> wrote:
> 
> I am using LUA to POST to a webservice and i receive a json string . How do i convert this string to a json object ?
> 
> -- 
> Ashwin Kumar Rath
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 royj at yandex.ru  Wed Feb 17 16:27:09 2016
From: royj at yandex.ru (royj at yandex.ru)
Date: Wed, 17 Feb 2016 16:27:09 +0300
Subject: [Freeswitch-users] hiredis: realm must be defined
In-Reply-To: <511661455627517@web2g.yandex.ru>
References: <511661455627517@web2g.yandex.ru>
Message-ID: <1070101455715629@web17o.yandex.ru>

Some strange for me really.
There is <profile name="myrealm"> in hiredis.conf.xml.
Also an application 'limit' with hiredis backend increments count on redis db, but after hangup does not decrease count (i look at ngrep on redis db host)

Is anybody used api limit_usage with hiredis backend?

16.02.2016, 16:02, "royj at yandex.ru" <royj at yandex.ru>:
> Hi, all
> Can anybody clarify what mean 'hiredis: realm must be defined'? Where is the misconfiguration?
> I am trying to use hiredis backend for limits and call api like 'api limit_usage hiredis myrealm id', got api/response '-1' and FreeSWITCH says in log 'hiredis: realm must be defined'.
> Module mod_hiredis loaded and exists.
> For hash backend works fine.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 asilva at wirelessmundi.com  Wed Feb 17 16:31:28 2016
From: asilva at wirelessmundi.com (Antonio Silva)
Date: Wed, 17 Feb 2016 14:31:28 +0100
Subject: [Freeswitch-users] Disable PRACK messages
Message-ID: <56C47630.7070402@wirelessmundi.com>

Hi,

Is it possible to disable or not to send prack messages?

I've set in the sip profile the option enable-100rel=false, but FS still 
sends the prack message...

My problem is that after receive the 180 from the remote side, FS sends 
the prack message and  the remote side reply with  SIP/2.0 481 
Call/Transaction Does Not Exist, This make FS to cancel the call.

I know that this is an issue on the remote side... but i want to avoid 
this message if possible.


Full sip trace in https://pastebin.freeswitch.org/24562


Thanks,
Ant?nio


From brian at freeswitch.org  Wed Feb 17 17:31:11 2016
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Feb 2016 08:31:11 -0600
Subject: [Freeswitch-users] no audio after hold
In-Reply-To: <CAK+V7RygPe5Bi-xzabjSdgovbvwo++k=X+FwZ0nqNQaeGkKYFQ@mail.gmail.com>
References: <CAK+V7RxSdxaSLAE3iwUqSpQkTorED0X24PpG0YL3tEbOgah3bw@mail.gmail.com>
	<CALXCt0rFM5xkK5JiwcqE5pj2pgFVkTaKZ8-qxcuqY4N1GGJLrA@mail.gmail.com>
	<CAEJMVkCRrBU8DC=PFua2yb_N48-xBR3r1v9YSaCG5Ss7sV4=_w@mail.gmail.com>
	<CAK+V7RygPe5Bi-xzabjSdgovbvwo++k=X+FwZ0nqNQaeGkKYFQ@mail.gmail.com>
Message-ID: <CAEJMVkDMFN3hPgSujiSvhk6ZpS-H+jmod468H9YKrW+XH7M+pQ@mail.gmail.com>

bug reports go on JIRA please.

On Wed, Feb 17, 2016 at 4:03 AM, Nick 'tarantul' Novikov <tarantul at gmail.com
> wrote:

> Hello!
>
> Log file is large, I can't paste it in FS pastebin.
> You may download gzip log file here
> https://yadi.sk/d/pLbEHi_LoyWsS
>
> On Tue, Feb 16, 2016 at 7:11 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Your logs are incomplete, need full debug logs.  Please don't filter them
>> excessively and try on master please.
>>
>> On Tue, Feb 16, 2016 at 10:06 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>> wrote:
>>
>>> Please pastebin a complete sip trace (from fs-cli: sofia global siptrace
>>> on) and then put here the link to pastebin.
>>> Il 16/Feb/2016 16:29, "Nick 'tarantul' Novikov" <tarantul at gmail.com> ha
>>> scritto:
>>>
>>>> Hello
>>>>
>>>> I have some problem with sound after hold usage. Sometime the sound
>>>> disappear after callcenter agent shift unhold.For callcenter agents we use
>>>> sip.js (version 0.7.2).
>>>> In freeswitch logs I see strange lines (
>>>> https://pastebin.freeswitch.org/24557).
>>>> Can anyone explain, what happened with audio and how to fix it?
>>>>
>>>>
>>>> --
>>>> tarantul
>>>> Dios es Amor
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>> 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
>>
>
>
>
> --
> tarantul
> Dios es Amor
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

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  Wed Feb 17 17:31:42 2016
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Feb 2016 08:31:42 -0600
Subject: [Freeswitch-users] early media after bridge
In-Reply-To: <28121455707236@web17o.yandex.ru>
References: <28121455707236@web17o.yandex.ru>
Message-ID: <CAEJMVkBwGAaZk47qvJhQDjF=Y01XjwJv3Q+t6ZrGEdRHqM3rpw@mail.gmail.com>

It SHOULD already be doing this unless you're ignoring early media on that
oubound leg.

On Wed, Feb 17, 2016 at 5:07 AM, <royj at yandex.ru> wrote:

> Hi, all
> Is there any chance to use the early media from the b-leg as ringback to
> the a-leg when a-leg is answered for play_and_get_digits for example?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

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 royj at yandex.ru  Wed Feb 17 17:57:19 2016
From: royj at yandex.ru (royj at yandex.ru)
Date: Wed, 17 Feb 2016 17:57:19 +0300
Subject: [Freeswitch-users] early media after bridge
In-Reply-To: <CAEJMVkBwGAaZk47qvJhQDjF=Y01XjwJv3Q+t6ZrGEdRHqM3rpw@mail.gmail.com>
References: <28121455707236@web17o.yandex.ru>
	<CAEJMVkBwGAaZk47qvJhQDjF=Y01XjwJv3Q+t6ZrGEdRHqM3rpw@mail.gmail.com>
Message-ID: <1543381455721039@web22g.yandex.ru>

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From tarantul at gmail.com  Wed Feb 17 23:32:37 2016
From: tarantul at gmail.com (Nick 'tarantul' Novikov)
Date: Wed, 17 Feb 2016 23:32:37 +0300
Subject: [Freeswitch-users] no audio after hold
In-Reply-To: <CAEJMVkDMFN3hPgSujiSvhk6ZpS-H+jmod468H9YKrW+XH7M+pQ@mail.gmail.com>
References: <CAK+V7RxSdxaSLAE3iwUqSpQkTorED0X24PpG0YL3tEbOgah3bw@mail.gmail.com>
	<CALXCt0rFM5xkK5JiwcqE5pj2pgFVkTaKZ8-qxcuqY4N1GGJLrA@mail.gmail.com>
	<CAEJMVkCRrBU8DC=PFua2yb_N48-xBR3r1v9YSaCG5Ss7sV4=_w@mail.gmail.com>
	<CAK+V7RygPe5Bi-xzabjSdgovbvwo++k=X+FwZ0nqNQaeGkKYFQ@mail.gmail.com>
	<CAEJMVkDMFN3hPgSujiSvhk6ZpS-H+jmod468H9YKrW+XH7M+pQ@mail.gmail.com>
Message-ID: <CAK+V7Rwfy3+fRsr49dmbfWeZ6fDhiaDdJMWXx5T6KD1XoJo7fA@mail.gmail.com>

Hello!

Done.
https://freeswitch.org/jira/browse/FS-8840

On Wed, Feb 17, 2016 at 5:31 PM, Brian West <brian at freeswitch.org> wrote:

> bug reports go on JIRA please.
>
> On Wed, Feb 17, 2016 at 4:03 AM, Nick 'tarantul' Novikov <
> tarantul at gmail.com> wrote:
>
>> Hello!
>>
>> Log file is large, I can't paste it in FS pastebin.
>> You may download gzip log file here
>> https://yadi.sk/d/pLbEHi_LoyWsS
>>
>> On Tue, Feb 16, 2016 at 7:11 PM, Brian West <brian at freeswitch.org> wrote:
>>
>>> Your logs are incomplete, need full debug logs.  Please don't filter
>>> them excessively and try on master please.
>>>
>>> On Tue, Feb 16, 2016 at 10:06 AM, Giovanni Maruzzelli <gmaruzz at gmail.com
>>> > wrote:
>>>
>>>> Please pastebin a complete sip trace (from fs-cli: sofia global
>>>> siptrace on) and then put here the link to pastebin.
>>>> Il 16/Feb/2016 16:29, "Nick 'tarantul' Novikov" <tarantul at gmail.com>
>>>> ha scritto:
>>>>
>>>>> Hello
>>>>>
>>>>> I have some problem with sound after hold usage. Sometime the sound
>>>>> disappear after callcenter agent shift unhold.For callcenter agents we use
>>>>> sip.js (version 0.7.2).
>>>>> In freeswitch logs I see strange lines (
>>>>> https://pastebin.freeswitch.org/24557).
>>>>> Can anyone explain, what happened with audio and how to fix it?
>>>>>
>>>>>
>>>>> --
>>>>> tarantul
>>>>> Dios es Amor
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>>
>>> 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
>>>
>>
>>
>>
>> --
>> tarantul
>> Dios es Amor
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> 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
>



-- 
tarantul
Dios es Amor
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From DEdwards at vertical.com  Thu Feb 18 00:16:47 2016
From: DEdwards at vertical.com (Dan Edwards)
Date: Wed, 17 Feb 2016 21:16:47 +0000
Subject: [Freeswitch-users] Can FS offer both internal & external IP address?
Message-ID: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C997FA45@PHXEX2.vertical.com>

I'm running FS behind Nginx, so I explicitly removed the local IP address from localnet.auto. This forces FS to always offer 'ext-rtp-ip'. Outside users work fine in that scenario.

Internal users also work, as long as we turn on NAT reflection in our router. I'm looking for a way to not require that router change.

If I can get FS to offer up both it's internal IP and 'ext-rtp-ip' as candidates in the initial INVITE, the browser could then use ICE to determine which IP address was correct. 

Is there a way to offer both and allow ICE on the client side to figure out which one to use?

Thanks,
Dan


From brian at freeswitch.org  Thu Feb 18 00:34:04 2016
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Feb 2016 15:34:04 -0600
Subject: [Freeswitch-users] Can FS offer both internal & external IP
	address?
In-Reply-To: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C997FA45@PHXEX2.vertical.com>
References: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C997FA45@PHXEX2.vertical.com>
Message-ID: <CAEJMVkCS556i8aWABomO4wz-XR93itqFfXOgCZpoD=09af4v+w@mail.gmail.com>

Actually no, what you do in this case is set the local-network-acl, it will
then know which one to use based on the request.

On Wed, Feb 17, 2016 at 3:16 PM, Dan Edwards <DEdwards at vertical.com> wrote:

> I'm running FS behind Nginx, so I explicitly removed the local IP address
> from localnet.auto. This forces FS to always offer 'ext-rtp-ip'. Outside
> users work fine in that scenario.
>
> Internal users also work, as long as we turn on NAT reflection in our
> router. I'm looking for a way to not require that router change.
>
> If I can get FS to offer up both it's internal IP and 'ext-rtp-ip' as
> candidates in the initial INVITE, the browser could then use ICE to
> determine which IP address was correct.
>
> Is there a way to offer both and allow ICE on the client side to figure
> out which one to use?
>
> Thanks,
> Dan
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

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 DEdwards at vertical.com  Thu Feb 18 00:45:05 2016
From: DEdwards at vertical.com (Dan Edwards)
Date: Wed, 17 Feb 2016 21:45:05 +0000
Subject: [Freeswitch-users] Can FS offer both internal & external
	IP	address?
In-Reply-To: <CAEJMVkCS556i8aWABomO4wz-XR93itqFfXOgCZpoD=09af4v+w@mail.gmail.com>
References: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C997FA45@PHXEX2.vertical.com>
	<CAEJMVkCS556i8aWABomO4wz-XR93itqFfXOgCZpoD=09af4v+w@mail.gmail.com>
Message-ID: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C997FA68@PHXEX2.vertical.com>

?based on the request? is what?s actually causing the problem. Since the requests come through a proxy, every ?remote_ip? is always the local IP, so FS cannot make a determination based on that. I added a deny to the local IP which caused FS to offer the external IP and we can make that work if our customers turn on the NAT reflection.

As a test, I modified switch_core_media.c:generate_m to throw in the local IP and that did fix our issue.  I was hoping for a configuration option.

Thanks,
Dan


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Wednesday, February 17, 2016 4:34 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Can FS offer both internal & external IP address?

Actually no, what you do in this case is set the local-network-acl, it will then know which one to use based on the request.

On Wed, Feb 17, 2016 at 3:16 PM, Dan Edwards <DEdwards at vertical.com<mailto:DEdwards at vertical.com>> wrote:
I'm running FS behind Nginx, so I explicitly removed the local IP address from localnet.auto. This forces FS to always offer 'ext-rtp-ip'. Outside users work fine in that scenario.

Internal users also work, as long as we turn on NAT reflection in our router. I'm looking for a way to not require that router change.

If I can get FS to offer up both it's internal IP and 'ext-rtp-ip' as candidates in the initial INVITE, the browser could then use ICE to determine which IP address was correct.

Is there a way to offer both and allow ICE on the client side to figure out which one to use?

Thanks,
Dan

_________________________________________________________________________
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



--

Brian West
brian at freeswitch.org<mailto:brian at freeswitch.org>

[http://billing.freeswitch.org/templates/default/img/whmcslogo.png]

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

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 ahabiba at gmail.com  Thu Feb 18 00:54:36 2016
From: ahabiba at gmail.com (Ahmed Habiba)
Date: Thu, 18 Feb 2016 00:54:36 +0300
Subject: [Freeswitch-users] Can FS offer both internal & external IP address?
In-Reply-To: <mailman.2301.1455743839.678.freeswitch-users@lists.freeswitch.org>
References: <mailman.2301.1455743839.678.freeswitch-users@lists.freeswitch.org>
Message-ID: <59068F3A-AFE8-463F-AE57-7E91E10A673A@gmail.com>

It is all about how you want things to run, FS allow to have multiple profile each of them can work on different ip considering that all users can call each other even they are registered on different IPs

here is below and example:

profile1:
     Name?> Internal
     IP      ?> 192.168.1.1
     SIP Port ?> 5060
 Sip_profile.xml:    <param name="ext-rtp-ip" value=?192.168.1.1"/>
  <param name="ext-sip-ip" value="192.168.1.1?/>


Profile2:
    Name ?> internal_nat
    IP       ?> 192.168.1.1
    SIP Port ?> 5065  ?>(Port Forwarded on Router to ?> RealIP:5060)
Sip_profile.xml  <param name="ext-rtp-ip" value="auto-nat"/>
  <param name="ext-sip-ip" value="auto-nat"/>

However of the above configuration still both internal and external will be able to call each other based on the below dial string , the ?*/? assure that users from different profile can call each other 

   conf/directory/default.xml :   <param name="dial-string" value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(*/${dialed_user}@${dialed_domain})}"/>



> On Feb 18, 2016, at 12:17 AM, freeswitch-users-request at lists.freeswitch.org wrote:
> 
> From: Dan Edwards <DEdwards at vertical.com <mailto:DEdwards at vertical.com>>
> Subject: [Freeswitch-users] Can FS offer both internal & external IP address?
> Date: February 18, 2016 at 12:16:47 AM GMT+3
> To: "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>>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> 
> 
> I'm running FS behind Nginx, so I explicitly removed the local IP address from localnet.auto. This forces FS to always offer 'ext-rtp-ip'. Outside users work fine in that scenario.
> 
> Internal users also work, as long as we turn on NAT reflection in our router. I'm looking for a way to not require that router change.
> 
> If I can get FS to offer up both it's internal IP and 'ext-rtp-ip' as candidates in the initial INVITE, the browser could then use ICE to determine which IP address was correct. 
> 
> Is there a way to offer both and allow ICE on the client side to figure out which one to use?
> 
> Thanks,
> Dan
> 

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From jprangi at gmail.com  Thu Feb 18 01:01:30 2016
From: jprangi at gmail.com (Jai Rangi)
Date: Wed, 17 Feb 2016 14:01:30 -0800
Subject: [Freeswitch-users] Oneway audio issues in freeswitch
In-Reply-To: <67F80D46-E511-459C-A72B-E966671D4AB3@dchorton.com>
References: <CANw+5nWmd_j=eUz7OGoH=5Wb3g43VOS_XF_dZVkuFRMRUtfZ2Q@mail.gmail.com>
	<CABMX_w_NWvQ5ZUdxtYHnua4FD9i4hYDt3YeP-X+43XYi4cb=8A@mail.gmail.com>
	<CAFLo2QzdR1YBLv6-y5hs=juyE5N2uZw+N5dCiyq4S3Hbgkb2eg@mail.gmail.com>
	<CANw+5nUWV3dmJGUtDgLBSaUuGufcXskoYNyj6FJcmd+yRs184g@mail.gmail.com>
	<CAPcNkdfppepyyC+OLTr4OFyhFWhP9MjWR0WUsraU-ABrcJLY1Q@mail.gmail.com>
	<CAPfjz3jwe74Vvz1515TQxOJdAYsNuRXA_CTtzTPfG9EjuXXhEA@mail.gmail.com>
	<67F80D46-E511-459C-A72B-E966671D4AB3@dchorton.com>
Message-ID: <CAPfjz3j7nvH2GFZatjndU7X9B+8r18XY5L3u62_CY4MXd1XzNg@mail.gmail.com>

Hello Dave,
Sorry for the delay, have been out of office.

Attached the the debug log file, with siptrace on. This is coming from
default configuration. I can confirm the issue happens only when this line
is commented out.

    <param name="auto-jitterbuffer-msec" value="60"/>

Not sure if this qualifies for bug, I can open a ticket in Jira.

Thank you,

-Jai


On Sat, Feb 13, 2016 at 7:49 PM, Dave Horton <dave at dchorton.com> wrote:

> Redo with sofia global siptrace on so we can see the SIP messaging
> On Feb 13, 2016, at 12:45 PM, Jai Rangi <jprangi at didforsale.com> wrote:
>
> Hello Ken,
> Thank for look in this. Attached are debug logs. SIP Traces were not
> molested, except the public IPs were changed. As of writing of this email,
> the issue is isolated to 1.6.x.
> Not sure if anyone else has tested this on latest version. But easy to
> reproduce. Just download grandstream Wave, available to IOS and Andriod and
> try to call any extension directly. Curious to see if any one can come with
> different result.
>
> *Jai Rangi*
> Cebod Technologies LLC dba DIDforSale/Cebod Telecom
> O 949-471-0102 <O%20949-471-0101> | C 949-419-7634 <C%20949-742-2666> | F
> 949-269-0449 / 949-232-1410 | jprangi at didforsale.com  www.cebod.com |
> www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA 92626  |
>
>
>
>
>
>
> On Thu, Feb 11, 2016 at 8:36 AM, Ken Rice <krice at freeswitch.org> wrote:
>
>> without logs of a call doing this at debug level with a complete
>> unmolested sip trace in line its a little hard to speculate whats going on
>> here
>>
>> On Thu, Feb 11, 2016 at 10:28 AM, mohammed shafeeque <shafeeq.v at gmail.com
>> > wrote:
>>
>>> Surprised that no one else experienced this problem. Can anyone give any
>>> hint. Really Dont want to move back to 1.4.x
>>>
>>> On Thu, Feb 11, 2016 at 7:44 AM, Luis Daniel Lucio Quiroz <
>>> luis.daniel.lucio at gmail.com> wrote:
>>>
>>>> As a rule of dumb, try turning on rport
>>>> Le 10 f?vr. 2016 8:55 PM, "?talo Rossi" <italo at freeswitch.org> a
>>>> ?crit :
>>>>
>>>>> You need to look at the sip signaling to see what's going on
>>>>>
>>>>> On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque <
>>>>> shafeeq.v at gmail.com> wrote:
>>>>>
>>>>>> Hello All
>>>>>>
>>>>>> We are getting one way audio issues with some softphones and
>>>>>> grandstream phones behind nat registerd to our freeswitch server.
>>>>>>
>>>>>> Here is scenario:
>>>>>> Grandstream call any extensions (one way audio)
>>>>>> Any extension call Grandstream ( Audio works just fine)
>>>>>>
>>>>>> We have tried multiple softphones and the result is same.
>>>>>>
>>>>>> Everything was working fine with 1.4.18 and 1.6.2. We were having
>>>>>> DTMF issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue
>>>>>> started with an upgrade to freeswitch.
>>>>>>
>>>>>> Any help or hint will be much appreciated.
>>>>>>
>>>>>> Thank you,
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> ?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
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> <freeswitch.log.goodcall.txt><freeswitch.log.badcall.txt>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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2016-02-17 13:47:25.637398 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/Master.csv
2016-02-17 13:47:25.637398 [NOTICE] mod_logfile.c:213 New log started.
recv 864 bytes from udp/[66.166.9.194]:39164 at 13:47:27.794790:
   ------------------------------------------------------------------------
   INVITE sip:1001 at 216.209.211.222 SIP/2.0
   Via: SIP/2.0/UDP 10.0.1.12:40886;branch=z9hG4bK281160191;rport
   From: <sip:1000 at 216.209.211.222>;tag=243846933
   To: <sip:1001 at 216.209.211.222>
   Call-ID: 930817686-40886-5 at BA.A.B.BC
   CSeq: 40 INVITE
   Contact: <sip:1000 at 10.0.1.12:40886>
   Max-Forwards: 70
   User-Agent: Grandstream Wave/IOS 1.0.19
   Privacy: none
   P-Preferred-Identity: <sip:1000 at 216.209.211.222>
   Supported: replaces, path, timer, eventlist
   Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
   Content-Type: application/sdp
   Accept: application/sdp, application/dtmf-relay
   Content-Length:   230
   
   v=0
   o=1000 8000 8000 IN IP4 10.0.1.12
   s=SIP Call
   c=IN IP4 10.0.1.12
   t=0 0
   m=audio 10270 RTP/AVP 0 8 101
   a=sendrecv
   a=rtpmap:0 PCMU/8000
   a=ptime:20
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   ------------------------------------------------------------------------
send 310 bytes to udp/[66.166.9.194]:39164 at 13:47:27.795559:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.0.1.12:40886;branch=z9hG4bK281160191;rport=39164;received=66.166.9.194
   From: <sip:1000 at 216.209.211.222>;tag=243846933
   To: <sip:1001 at 216.209.211.222>
   Call-ID: 930817686-40886-5 at BA.A.B.BC
   CSeq: 40 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit
   Content-Length: 0
   
   ------------------------------------------------------------------------
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.777377 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/1000 at 216.209.211.222 [4008c053-8780-4987-b84f-010b67013bda]
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.777377 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1000 at 216.209.211.222) Running State Change CS_NEW
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.777377 [DEBUG] sofia.c:9248 sofia/internal/1000 at 216.209.211.222 receiving invite from 66.166.9.194:39164 version: 1.6.6  64bit
2016-02-17 13:47:27.777377 [DEBUG] sofia.c:9415 IP 66.166.9.194 Rejected by acl "domains". Falling back to Digest auth.
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.797388 [DEBUG] switch_core_state_machine.c:492 (sofia/internal/1000 at 216.209.211.222) State NEW
send 814 bytes to udp/[66.166.9.194]:39164 at 13:47:27.798293:
   ------------------------------------------------------------------------
   SIP/2.0 407 Proxy Authentication Required
   Via: SIP/2.0/UDP 10.0.1.12:40886;branch=z9hG4bK281160191;rport=39164;received=66.166.9.194
   From: <sip:1000 at 216.209.211.222>;tag=243846933
   To: <sip:1001 at 216.209.211.222>;tag=Drpp4yp9j1X2m
   Call-ID: 930817686-40886-5 at BA.A.B.BC
   CSeq: 40 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Proxy-Authenticate: Digest realm="216.209.211.222", nonce="696ccc87-c214-4a8a-870f-7e3d69ca016d", algorithm=MD5, qop="auth"
   Content-Length: 0
   
   ------------------------------------------------------------------------
2016-02-17 13:47:27.797388 [DEBUG] sofia.c:2147 detaching session 4008c053-8780-4987-b84f-010b67013bda
recv 267 bytes from udp/[66.166.9.194]:39164 at 13:47:27.843925:
   ------------------------------------------------------------------------
   ACK sip:1001 at 216.209.211.222 SIP/2.0
   Via: SIP/2.0/UDP 10.0.1.12:40886;branch=z9hG4bK281160191;rport
   From: <sip:1000 at 216.209.211.222>;tag=243846933
   To: <sip:1001 at 216.209.211.222>;tag=Drpp4yp9j1X2m
   Call-ID: 930817686-40886-5 at BA.A.B.BC
   CSeq: 40 ACK
   Content-Length: 0
   
   ------------------------------------------------------------------------
recv 1111 bytes from udp/[66.166.9.194]:39164 at 13:47:27.860855:
   ------------------------------------------------------------------------
   INVITE sip:1001 at 216.209.211.222 SIP/2.0
   Via: SIP/2.0/UDP 10.0.1.12:40886;branch=z9hG4bK1578402832;rport
   From: <sip:1000 at 216.209.211.222>;tag=243846933
   To: <sip:1001 at 216.209.211.222>
   Call-ID: 930817686-40886-5 at BA.A.B.BC
   CSeq: 41 INVITE
   Contact: <sip:1000 at 10.0.1.12:40886>
   Proxy-Authorization: Digest username="1000", realm="216.209.211.222", nonce="696ccc87-c214-4a8a-870f-7e3d69ca016d", uri="sip:1001 at 216.209.211.222", response="73ea6df6242e1f2e5cc6e4d6bad9be3e", algorithm=MD5, cnonce="15059615", qop=auth, nc=00000006
   Max-Forwards: 70
   User-Agent: Grandstream Wave/IOS 1.0.19
   Privacy: none
   P-Preferred-Identity: <sip:1000 at 216.209.211.222>
   Supported: replaces, path, timer, eventlist
   Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
   Content-Type: application/sdp
   Accept: application/sdp, application/dtmf-relay
   Content-Length:   230
   
   v=0
   o=1000 8000 8000 IN IP4 10.0.1.12
   s=SIP Call
   c=IN IP4 10.0.1.12
   t=0 0
   m=audio 10270 RTP/AVP 0 8 101
   a=sendrecv
   a=rtpmap:0 PCMU/8000
   a=ptime:20
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   ------------------------------------------------------------------------
send 311 bytes to udp/[66.166.9.194]:39164 at 13:47:27.862041:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.0.1.12:40886;branch=z9hG4bK1578402832;rport=39164;received=66.166.9.194
   From: <sip:1000 at 216.209.211.222>;tag=243846933
   To: <sip:1001 at 216.209.211.222>
   Call-ID: 930817686-40886-5 at BA.A.B.BC
   CSeq: 41 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit
   Content-Length: 0
   
   ------------------------------------------------------------------------
2016-02-17 13:47:27.857368 [DEBUG] sofia.c:2255 Re-attaching to session 4008c053-8780-4987-b84f-010b67013bda
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] sofia.c:9248 sofia/internal/1000 at 216.209.211.222 receiving invite from 66.166.9.194:39164 version: 1.6.6  64bit
2016-02-17 13:47:27.877397 [DEBUG] sofia.c:9415 IP 66.166.9.194 Rejected by acl "domains". Falling back to Digest auth.
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] sofia.c:10549 Setting NAT mode based on nat.auto
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] sofia.c:6760 Channel sofia/internal/1000 at 216.209.211.222 entering state [received][100]
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] sofia.c:6770 Remote SDP:
4008c053-8780-4987-b84f-010b67013bda v=0
4008c053-8780-4987-b84f-010b67013bda o=1000 8000 8000 IN IP4 10.0.1.12
4008c053-8780-4987-b84f-010b67013bda s=SIP Call
4008c053-8780-4987-b84f-010b67013bda c=IN IP4 10.0.1.12
4008c053-8780-4987-b84f-010b67013bda t=0 0
4008c053-8780-4987-b84f-010b67013bda m=audio 10270 RTP/AVP 0 8 101
4008c053-8780-4987-b84f-010b67013bda a=rtpmap:0 PCMU/8000
4008c053-8780-4987-b84f-010b67013bda a=rtpmap:8 PCMA/8000
4008c053-8780-4987-b84f-010b67013bda a=rtpmap:101 telephone-event/8000
4008c053-8780-4987-b84f-010b67013bda a=fmtp:101 0-15
4008c053-8780-4987-b84f-010b67013bda a=ptime:20
4008c053-8780-4987-b84f-010b67013bda 
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] sofia.c:7125 (sofia/internal/1000 at 216.209.211.222) State Change CS_NEW -> CS_INIT
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1000 at 216.209.211.222) Running State Change CS_INIT
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/1000 at 216.209.211.222) State INIT
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] mod_sofia.c:88 sofia/internal/1000 at 216.209.211.222 SOFIA INIT
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_core_state_machine.c:40 sofia/internal/1000 at 216.209.211.222 Standard INIT
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/1000 at 216.209.211.222) State Change CS_INIT -> CS_ROUTING
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/1000 at 216.209.211.222) State INIT going to sleep
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1000 at 216.209.211.222) Running State Change CS_ROUTING
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_channel.c:2247 (sofia/internal/1000 at 216.209.211.222) Callstate Change DOWN -> RINGING
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/1000 at 216.209.211.222) State ROUTING
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] mod_sofia.c:141 sofia/internal/1000 at 216.209.211.222 SOFIA ROUTING
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_core_state_machine.c:166 sofia/internal/1000 at 216.209.211.222 Standard ROUTING
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1001 in context default
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->unloop] continue=false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->tod_example] continue=true
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Date/Time Match (PASS) [tod_example] break=on-false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action set(open=true)
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->holiday_example] continue=true
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Date/TimeMatch (FAIL) [holiday_example] break=on-false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->global-intercept] continue=false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [global-intercept] destination_number(1001) =~ /^886$/ break=on-false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->group-intercept] continue=false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [group-intercept] destination_number(1001) =~ /^\*8$/ break=on-false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->intercept-ext] continue=false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [intercept-ext] destination_number(1001) =~ /^\*\*(\d+)$/ break=on-false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->redial] continue=false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [redial] destination_number(1001) =~ /^(redial|870)$/ break=on-false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->global] continue=true
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [global] ${default_password}(qwerty at 123$) =~ /^1234$/ break=never
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [global] ${rtp_has_crypto}() =~ /^(AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8|AES_CM_256_HMAC_SHA1_80|AES_CM_192_HMAC_SHA1_80|AES_CM_128_HMAC_SHA1_80|AES_CM_256_HMAC_SHA1_32|AES_CM_192_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_32|AES_CM_128_NULL_AUTH)$/ break=never
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (PASS) [global] ${endpoint_disposition}(DELAYED NEGOTIATION) =~ /^(DELAYED NEGOTIATION)/ break=on-false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [global] ${switch_r_sdp}(v=0
4008c053-8780-4987-b84f-010b67013bda o=1000 8000 8000 IN IP4 10.0.1.12
4008c053-8780-4987-b84f-010b67013bda s=SIP Call
4008c053-8780-4987-b84f-010b67013bda c=IN IP4 10.0.1.12
4008c053-8780-4987-b84f-010b67013bda t=0 0
4008c053-8780-4987-b84f-010b67013bda m=audio 10270 RTP/AVP 0 8 101
4008c053-8780-4987-b84f-010b67013bda a=rtpmap:0 PCMU/8000
4008c053-8780-4987-b84f-010b67013bda a=rtpmap:8 PCMA/8000
4008c053-8780-4987-b84f-010b67013bda a=rtpmap:101 telephone-event/8000
4008c053-8780-4987-b84f-010b67013bda a=fmtp:101 0-15
4008c053-8780-4987-b84f-010b67013bda a=ptime:20
4008c053-8780-4987-b84f-010b67013bda ) =~ /(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)/ break=never
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Absolute Condition [global]
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action hash(insert/${domain_name}-last_dial/global/${uuid})
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->snom-demo-2] continue=false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [snom-demo-2] destination_number(1001) =~ /^9001$/ break=on-false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->snom-demo-1] continue=false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [snom-demo-1] destination_number(1001) =~ /^9000$/ break=on-false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->eavesdrop] continue=false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [eavesdrop] destination_number(1001) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->eavesdrop] continue=false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [eavesdrop] destination_number(1001) =~ /^779$/ break=on-false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->call_return] continue=false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [call_return] destination_number(1001) =~ /^\*69$|^869$|^lcr$/ break=on-false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->del-group] continue=false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [del-group] destination_number(1001) =~ /^80(\d{2})$/ break=on-false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->add-group] continue=false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [add-group] destination_number(1001) =~ /^81(\d{2})$/ break=on-false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->call-group-simo] continue=false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [call-group-simo] destination_number(1001) =~ /^82(\d{2})$/ break=on-false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->call-group-order] continue=false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [call-group-order] destination_number(1001) =~ /^83(\d{2})$/ break=on-false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->extension-intercom] continue=false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [extension-intercom] destination_number(1001) =~ /^8(10[01][0-9])$/ break=on-false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->Local_Extension] continue=false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (PASS) [Local_Extension] destination_number(1001) =~ /^(10[01][0-9])$/ break=on-false
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action export(dialed_extension=1001)
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action bind_meta_app(1 b s execute_extension::dx XML features)
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/var/lib/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav)
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action bind_meta_app(3 b s execute_extension::cf XML features)
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action bind_meta_app(4 b s execute_extension::att_xfer XML features)
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action set(ringback=${us-ring})
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action set(transfer_ringback=local_stream://moh)
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action set(call_timeout=30)
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action set(hangup_after_bridge=true)
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action set(continue_on_fail=true)
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number})
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)})
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action hash(insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid})
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action hash(insert/${domain_name}-last_dial_ext/global/${uuid})
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action bridge(user/${dialed_extension}@${domain_name})
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action answer()
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action sleep(1000)
4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension})
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_core_state_machine.c:216 (sofia/internal/1000 at 216.209.211.222) State Change CS_ROUTING -> CS_EXECUTE
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/1000 at 216.209.211.222) State ROUTING going to sleep
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1000 at 216.209.211.222) Running State Change CS_EXECUTE
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_core_state_machine.c:539 (sofia/internal/1000 at 216.209.211.222) State EXECUTE
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] mod_sofia.c:196 sofia/internal/1000 at 216.209.211.222 SOFIA EXECUTE
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_core_state_machine.c:258 sofia/internal/1000 at 216.209.211.222 Standard EXECUTE
4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 set(open=true)
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1000 at 216.209.211.222 [open]=[true]
4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 hash(insert/216.209.211.222-spymap/1000/4008c053-8780-4987-b84f-010b67013bda)
4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 hash(insert/216.209.211.222-last_dial/1000/1001)
4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 hash(insert/216.209.211.222-last_dial/global/4008c053-8780-4987-b84f-010b67013bda)
4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 export(RFC2822_DATE=Wed, 17 Feb 2016 13:47:27 -0800)
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [RFC2822_DATE]=[Wed, 17 Feb 2016 13:47:27 -0800]
4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 export(dialed_extension=1001)
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [dialed_extension]=[1001]
4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 bind_meta_app(1 b s execute_extension::dx XML features)
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *1 execute_extension::dx XML features
4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 bind_meta_app(2 b s record_session::/usr/local/freeswitch/var/lib/freeswitch/recordings/1000.2016-02-17-13-47-27.wav)
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *2 record_session::/usr/local/freeswitch/var/lib/freeswitch/recordings/1000.2016-02-17-13-47-27.wav
4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 bind_meta_app(3 b s execute_extension::cf XML features)
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *3 execute_extension::cf XML features
4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 bind_meta_app(4 b s execute_extension::att_xfer XML features)
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *4 execute_extension::att_xfer XML features
4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 set(ringback=%(2000,4000,440,480))
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1000 at 216.209.211.222 [ringback]=[%(2000,4000,440,480)]
4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 set(transfer_ringback=local_stream://moh)
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1000 at 216.209.211.222 [transfer_ringback]=[local_stream://moh]
4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 set(call_timeout=30)
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1000 at 216.209.211.222 [call_timeout]=[30]
4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 set(hangup_after_bridge=true)
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1000 at 216.209.211.222 [hangup_after_bridge]=[true]
4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 set(continue_on_fail=true)
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.897386 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1000 at 216.209.211.222 [continue_on_fail]=[true]
4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 hash(insert/216.209.211.222-call_return/1001/1000)
4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 hash(insert/216.209.211.222-last_dial_ext/1001/4008c053-8780-4987-b84f-010b67013bda)
4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 set(called_party_callgroup=techsupport)
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.897386 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1000 at 216.209.211.222 [called_party_callgroup]=[techsupport]
4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 hash(insert/216.209.211.222-last_dial_ext/techsupport/4008c053-8780-4987-b84f-010b67013bda)
4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 hash(insert/216.209.211.222-last_dial_ext/global/4008c053-8780-4987-b84f-010b67013bda)
4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 hash(insert/216.209.211.222-last_dial/techsupport/4008c053-8780-4987-b84f-010b67013bda)
4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 bridge(user/1001 at 216.209.211.222)
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.897386 [DEBUG] switch_channel.c:1247 sofia/internal/1000 at 216.209.211.222 EXPORTING[export_vars] [RFC2822_DATE]=[Wed, 17 Feb 2016 13:47:27 -0800] to event
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.897386 [DEBUG] switch_channel.c:1247 sofia/internal/1000 at 216.209.211.222 EXPORTING[export_vars] [dialed_extension]=[1001] to event
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.897386 [DEBUG] switch_ivr_originate.c:2128 Parsing global variables
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.897386 [DEBUG] switch_channel.c:1247 sofia/internal/1000 at 216.209.211.222 EXPORTING[export_vars] [RFC2822_DATE]=[Wed, 17 Feb 2016 13:47:27 -0800] to event
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.897386 [DEBUG] switch_channel.c:1247 sofia/internal/1000 at 216.209.211.222 EXPORTING[export_vars] [dialed_extension]=[1001] to event
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.897386 [DEBUG] switch_ivr_originate.c:2128 Parsing global variables
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/1001 at 10.0.1.2:40886 [15df2523-1e92-49ca-bf21-bbf07d51c29e]
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] mod_sofia.c:4776 (sofia/internal/1001 at 10.0.1.2:40886) State Change CS_NEW -> CS_INIT
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1001 at 10.0.1.2:40886) Running State Change CS_INIT
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/1001 at 10.0.1.2:40886) State INIT
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] mod_sofia.c:88 sofia/internal/1001 at 10.0.1.2:40886 SOFIA INIT
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] sofia_glue.c:1228 sip:1001 at 66.166.9.194:41958 Setting proxy route to sofia/internal/1001 at 10.0.1.2:40886
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] sofia_glue.c:1257 sofia/internal/1001 at 10.0.1.2:40886 sending invite version: 1.6.6  64bit
15df2523-1e92-49ca-bf21-bbf07d51c29e Local SDP:
15df2523-1e92-49ca-bf21-bbf07d51c29e v=0
15df2523-1e92-49ca-bf21-bbf07d51c29e o=FreeSWITCH 1455713039 1455713040 IN IP4 216.209.211.222
15df2523-1e92-49ca-bf21-bbf07d51c29e s=FreeSWITCH
15df2523-1e92-49ca-bf21-bbf07d51c29e c=IN IP4 216.209.211.222
15df2523-1e92-49ca-bf21-bbf07d51c29e t=0 0
15df2523-1e92-49ca-bf21-bbf07d51c29e m=audio 32608 RTP/AVP 0 8 101
15df2523-1e92-49ca-bf21-bbf07d51c29e a=rtpmap:0 PCMU/8000
15df2523-1e92-49ca-bf21-bbf07d51c29e a=rtpmap:8 PCMA/8000
15df2523-1e92-49ca-bf21-bbf07d51c29e a=rtpmap:101 telephone-event/8000
15df2523-1e92-49ca-bf21-bbf07d51c29e a=fmtp:101 0-16
15df2523-1e92-49ca-bf21-bbf07d51c29e a=ptime:20
15df2523-1e92-49ca-bf21-bbf07d51c29e a=sendrecv
15df2523-1e92-49ca-bf21-bbf07d51c29e 
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] switch_core_state_machine.c:40 sofia/internal/1001 at 10.0.1.2:40886 Standard INIT
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/1001 at 10.0.1.2:40886) State Change CS_INIT -> CS_ROUTING
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/1001 at 10.0.1.2:40886) State INIT going to sleep
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1001 at 10.0.1.2:40886) Running State Change CS_ROUTING
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/1001 at 10.0.1.2:40886) State ROUTING
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] mod_sofia.c:141 sofia/internal/1001 at 10.0.1.2:40886 SOFIA ROUTING
send 1185 bytes to udp/[66.166.9.194]:41958 at 13:47:27.912034:
   ------------------------------------------------------------------------
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/1001 at 10.0.1.2:40886) State Change CS_ROUTING -> CS_CONSUME_MEDIA
   INVITE sip:1001 at 10.0.1.2:40886 SIP/2.0
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/1001 at 10.0.1.2:40886) State ROUTING going to sleep
   Via: SIP/2.0/UDP 216.209.211.222;rport;branch=z9hG4bK38Xat6Z71UHUN
   Route: <sip:1001 at 66.166.9.194:41958>
   Max-Forwards: 69
   From: "Extension 1000" <sip:1000 at 216.209.211.222>;tag=Fa977mrgDKa8B
   To: <sip:1001 at 10.0.1.2:40886>
   Call-ID: e124e149-5062-1234-9bb4-c6827b410b7a
   CSeq: 87536759 INVITE
   Contact: <sip:mod_sofia at 216.209.211.222:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Privacy: none
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 246
   X-FS-Support: update_display,send_info
   P-Asserted-Identity: "Extension 1000" <sip:1000 at 216.209.211.222>
   
   v=0
   o=FreeSWITCH 1455713039 1455713040 IN IP4 216.209.211.222
   s=FreeSWITCH
   c=IN IP4 216.209.211.222
   t=0 0
   m=audio 32608 RTP/AVP 0 8 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1001 at 10.0.1.2:40886) Running State Change CS_CONSUME_MEDIA
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] sofia.c:6760 Channel sofia/internal/1001 at 10.0.1.2:40886 entering state [calling][0]
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] switch_core_state_machine.c:551 (sofia/internal/1001 at 10.0.1.2:40886) State CONSUME_MEDIA
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] switch_core_state_machine.c:551 (sofia/internal/1001 at 10.0.1.2:40886) State CONSUME_MEDIA going to sleep
recv 450 bytes from udp/[66.166.9.194]:41958 at 13:47:28.028660:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 216.209.211.222;rport=5060;branch=z9hG4bK38Xat6Z71UHUN
   From: "Extension 1000" <sip:1000 at 216.209.211.222>;tag=Fa977mrgDKa8B
   To: <sip:1001 at 10.0.1.2:40886>
   Call-ID: e124e149-5062-1234-9bb4-c6827b410b7a
   CSeq: 87536759 INVITE
   Supported: replaces, path, eventlist
   User-Agent: Grandstream Wave/IOS 1.0.19
   Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
   Content-Length: 0
   
   ------------------------------------------------------------------------
recv 535 bytes from udp/[66.166.9.194]:41958 at 13:47:28.036530:
   ------------------------------------------------------------------------
   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP 216.209.211.222;rport=5060;branch=z9hG4bK38Xat6Z71UHUN
   From: "Extension 1000" <sip:1000 at 216.209.211.222>;tag=Fa977mrgDKa8B
   To: <sip:1001 at 10.0.1.2:40886>;tag=2072085022
   Call-ID: e124e149-5062-1234-9bb4-c6827b410b7a
   CSeq: 87536759 INVITE
   Contact: <sip:1001 at 10.0.1.2:40886>
   Supported: replaces, path, timer, eventlist
   User-Agent: Grandstream Wave/IOS 1.0.19
   Allow-Events: talk, hold
   Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
   Content-Length: 0
   
   ------------------------------------------------------------------------
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:28.037389 [DEBUG] sofia.c:6760 Channel sofia/internal/1001 at 10.0.1.2:40886 entering state [proceeding][180]
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:28.037389 [NOTICE] sofia.c:6862 Ring-Ready sofia/internal/1001 at 10.0.1.2:40886!
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:28.037389 [DEBUG] switch_channel.c:3340 (sofia/internal/1001 at 10.0.1.2:40886) Callstate Change DOWN -> RINGING
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [INFO] switch_ivr_originate.c:1216 Sending early media
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[opus:116:48000:20:0:1]
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1]
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[opus:116:48000:20:0:1]
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1]
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:4077 Set telephone-event payload to 101 at 8000
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:2906 Set Codec sofia/internal/1000 at 216.209.211.222 PCMU/8000 20 ms 160 samples 64000 bits 1 channels
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_codec.c:111 sofia/internal/1000 at 216.209.211.222 Original read codec set to PCMU:0
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:4429 Set telephone-event payload to 101 at 8000
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:4485 sofia/internal/1000 at 216.209.211.222 Set 2833 dtmf send payload to 101 recv payload to 101
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:6033 AUDIO RTP [sofia/internal/1000 at 216.209.211.222] 216.209.211.222 port 25166 -> 10.0.1.12 port 10270 codec: 0 ms: 20
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_rtp.c:3802 Starting timer [soft] 160 bytes per 20ms
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:1939 Setting Jitterbuffer to 60ms (3 frames) (50 max frames)
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:6332 sofia/internal/1000 at 216.209.211.222 Set 2833 dtmf send payload to 101
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:6339 sofia/internal/1000 at 216.209.211.222 Set 2833 dtmf receive payload to 101
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:6362 sofia/internal/1000 at 216.209.211.222 Set rtp dtmf delay to 40
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000 at 216.209.211.222!
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.057400 [DEBUG] switch_channel.c:3468 (sofia/internal/1000 at 216.209.211.222) Callstate Change RINGING -> EARLY
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.057400 [DEBUG] mod_sofia.c:2330 Ring SDP:
4008c053-8780-4987-b84f-010b67013bda v=0
4008c053-8780-4987-b84f-010b67013bda o=FreeSWITCH 1455720482 1455720483 IN IP4 216.209.211.222
4008c053-8780-4987-b84f-010b67013bda s=FreeSWITCH
4008c053-8780-4987-b84f-010b67013bda c=IN IP4 216.209.211.222
4008c053-8780-4987-b84f-010b67013bda t=0 0
4008c053-8780-4987-b84f-010b67013bda m=audio 25166 RTP/AVP 0 101
4008c053-8780-4987-b84f-010b67013bda a=rtpmap:0 PCMU/8000
4008c053-8780-4987-b84f-010b67013bda a=rtpmap:101 telephone-event/8000
4008c053-8780-4987-b84f-010b67013bda a=fmtp:101 0-16
4008c053-8780-4987-b84f-010b67013bda a=ptime:20
4008c053-8780-4987-b84f-010b67013bda a=sendrecv
4008c053-8780-4987-b84f-010b67013bda 
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.057400 [DEBUG] switch_ivr_originate.c:1274 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.057400 [DEBUG] switch_core_codec.c:221 sofia/internal/1000 at 216.209.211.222 Push codec L16:100
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.057400 [DEBUG] switch_ivr_originate.c:1343 Play Ringback Tone [%(2000,4000,440,480)]
send 1072 bytes to udp/[66.166.9.194]:39164 at 13:47:28.060325:
   ------------------------------------------------------------------------
   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP 10.0.1.12:40886;branch=z9hG4bK1578402832;rport=39164;received=66.166.9.194
   From: <sip:1000 at 216.209.211.222>;tag=243846933
   To: <sip:1001 at 216.209.211.222>;tag=e1FF6S7cgamNg
   Call-ID: 930817686-40886-5 at BA.A.B.BC
   CSeq: 41 INVITE
   Contact: <sip:1001 at 216.209.211.222:5060;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 222
   P-Asserted-Identity: "1001" <sip:1001 at 216.209.211.222>
   
   v=0
   o=FreeSWITCH 1455720482 1455720483 IN IP4 216.209.211.222
   s=FreeSWITCH
   c=IN IP4 216.209.211.222
   t=0 0
   m=audio 25166 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.077397 [DEBUG] sofia.c:6760 Channel sofia/internal/1000 at 216.209.211.222 entering state [early][183]
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.817431 [INFO] switch_rtp.c:6616 Auto Changing audio port from 10.0.1.12:10270 to 66.166.9.194:38124
recv 767 bytes from udp/[66.166.9.194]:41958 at 13:47:30.718270:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 216.209.211.222;rport=5060;branch=z9hG4bK38Xat6Z71UHUN
   From: "Extension 1000" <sip:1000 at 216.209.211.222>;tag=Fa977mrgDKa8B
   To: <sip:1001 at 10.0.1.2:40886>;tag=2072085022
   Call-ID: e124e149-5062-1234-9bb4-c6827b410b7a
   CSeq: 87536759 INVITE
   Contact: <sip:1001 at 10.0.1.2:40886>
   Supported: replaces, path, timer, eventlist
   User-Agent: Grandstream Wave/IOS 1.0.19
   Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
   Content-Type: application/sdp
   Content-Length:   228
   
   v=0
   o=1001 8001 8000 IN IP4 10.0.1.2
   s=SIP Call
   c=IN IP4 10.0.1.2
   t=0 0
   m=audio 35544 RTP/AVP 0 8 101
   a=sendrecv
   a=rtpmap:0 PCMU/8000
   a=ptime:20
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   ------------------------------------------------------------------------
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] sofia.c:6760 Channel sofia/internal/1001 at 10.0.1.2:40886 entering state [completing][200]
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] sofia.c:6770 Remote SDP:
15df2523-1e92-49ca-bf21-bbf07d51c29e v=0
15df2523-1e92-49ca-bf21-bbf07d51c29e o=1001 8001 8000 IN IP4 10.0.1.2
15df2523-1e92-49ca-bf21-bbf07d51c29e s=SIP Call
15df2523-1e92-49ca-bf21-bbf07d51c29e c=IN IP4 10.0.1.2
15df2523-1e92-49ca-bf21-bbf07d51c29e t=0 0
15df2523-1e92-49ca-bf21-bbf07d51c29e m=audio 35544 RTP/AVP 0 8 101
15df2523-1e92-49ca-bf21-bbf07d51c29e a=rtpmap:0 PCMU/8000
15df2523-1e92-49ca-bf21-bbf07d51c29e a=rtpmap:8 PCMA/8000
15df2523-1e92-49ca-bf21-bbf07d51c29e a=rtpmap:101 telephone-event/8000
15df2523-1e92-49ca-bf21-bbf07d51c29e a=fmtp:101 0-15
15df2523-1e92-49ca-bf21-bbf07d51c29e a=ptime:20
15df2523-1e92-49ca-bf21-bbf07d51c29e 
send 367 bytes to udp/[66.166.9.194]:41958 at 13:47:30.723049:
   ------------------------------------------------------------------------
   ACK sip:1001 at 10.0.1.2:40886 SIP/2.0
   Via: SIP/2.0/UDP 216.209.211.222;rport;branch=z9hG4bK4HQ3U1gBZ47DH
   Max-Forwards: 70
   From: "Extension 1000" <sip:1000 at 216.209.211.222>;tag=Fa977mrgDKa8B
   To: <sip:1001 at 10.0.1.2:40886>;tag=2072085022
   Call-ID: e124e149-5062-1234-9bb4-c6827b410b7a
   CSeq: 87536759 ACK
   Contact: <sip:mod_sofia at 216.209.211.222:5060>
   Content-Length: 0
   
   ------------------------------------------------------------------------
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] sofia.c:6760 Channel sofia/internal/1001 at 10.0.1.2:40886 entering state [ready][200]
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:4077 Set telephone-event payload to 101 at 8000
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:2906 Set Codec sofia/internal/1001 at 10.0.1.2:40886 PCMU/8000 20 ms 160 samples 64000 bits 1 channels
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_codec.c:111 sofia/internal/1001 at 10.0.1.2:40886 Original read codec set to PCMU:0
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:4429 Set telephone-event payload to 101 at 8000
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:4485 sofia/internal/1001 at 10.0.1.2:40886 Set 2833 dtmf send payload to 101 recv payload to 101
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:6033 AUDIO RTP [sofia/internal/1001 at 10.0.1.2:40886] 216.209.211.222 port 32608 -> 10.0.1.2 port 35544 codec: 0 ms: 20
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_rtp.c:3802 Starting timer [soft] 160 bytes per 20ms
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:1939 Setting Jitterbuffer to 60ms (3 frames) (50 max frames)
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:6332 sofia/internal/1001 at 10.0.1.2:40886 Set 2833 dtmf send payload to 101
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:6339 sofia/internal/1001 at 10.0.1.2:40886 Set 2833 dtmf receive payload to 101
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:6362 sofia/internal/1001 at 10.0.1.2:40886 Set rtp dtmf delay to 40
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [NOTICE] sofia.c:7724 Channel [sofia/internal/1001 at 10.0.1.2:40886] has been answered
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_channel.c:3767 (sofia/internal/1001 at 10.0.1.2:40886) Callstate Change RINGING -> ACTIVE
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:30.737383 [DEBUG] switch_core_codec.c:246 sofia/internal/1000 at 216.209.211.222 Restore previous codec PCMU:0.
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:30.737383 [DEBUG] mod_sofia.c:799 Local SDP sofia/internal/1000 at 216.209.211.222:
4008c053-8780-4987-b84f-010b67013bda v=0
4008c053-8780-4987-b84f-010b67013bda o=FreeSWITCH 1455720482 1455720484 IN IP4 216.209.211.222
4008c053-8780-4987-b84f-010b67013bda s=FreeSWITCH
4008c053-8780-4987-b84f-010b67013bda c=IN IP4 216.209.211.222
4008c053-8780-4987-b84f-010b67013bda t=0 0
4008c053-8780-4987-b84f-010b67013bda m=audio 25166 RTP/AVP 0 101
4008c053-8780-4987-b84f-010b67013bda a=rtpmap:0 PCMU/8000
4008c053-8780-4987-b84f-010b67013bda a=rtpmap:101 telephone-event/8000
4008c053-8780-4987-b84f-010b67013bda a=fmtp:101 0-16
4008c053-8780-4987-b84f-010b67013bda a=ptime:20
4008c053-8780-4987-b84f-010b67013bda a=sendrecv
4008c053-8780-4987-b84f-010b67013bda 
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:30.737383 [NOTICE] switch_ivr_originate.c:3550 Channel [sofia/internal/1000 at 216.209.211.222] has been answered
send 1094 bytes to udp/[66.166.9.194]:39164 at 13:47:30.743468:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 10.0.1.12:40886;branch=z9hG4bK1578402832;rport=39164;received=66.166.9.194
   From: <sip:1000 at 216.209.211.222>;tag=243846933
   To: <sip:1001 at 216.209.211.222>;tag=e1FF6S7cgamNg
   Call-ID: 930817686-40886-5 at BA.A.B.BC
   CSeq: 41 INVITE
   Contact: <sip:1001 at 216.209.211.222:5060;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Require: timer
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Session-Expires: 120;refresher=uac
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 222
   P-Asserted-Identity: "Outbound Call" <sip:1001 at 216.209.211.222>
   
   v=0
   o=FreeSWITCH 1455720482 1455720483 IN IP4 216.209.211.222
   s=FreeSWITCH
   c=IN IP4 216.209.211.222
   t=0 0
   m=audio 25166 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:30.737383 [DEBUG] switch_channel.c:3767 (sofia/internal/1000 at 216.209.211.222) Callstate Change EARLY -> ACTIVE
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:30.737383 [DEBUG] sofia.c:6760 Channel sofia/internal/1000 at 216.209.211.222 entering state [completed][200]
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:30.737383 [DEBUG] switch_ivr_originate.c:3608 Originate Resulted in Success: [sofia/internal/1001 at 10.0.1.2:40886]
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:30.737383 [DEBUG] switch_ivr_originate.c:3608 Originate Resulted in Success: [sofia/internal/1001 at 10.0.1.2:40886]
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.737383 [DEBUG] switch_core_media.c:9156 sofia/internal/1001 at 10.0.1.2:40886 PAUSE Jitterbuffer
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:30.737383 [DEBUG] switch_core_media.c:9156 sofia/internal/1000 at 216.209.211.222 PAUSE Jitterbuffer
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.737383 [DEBUG] switch_ivr_bridge.c:1591 (sofia/internal/1001 at 10.0.1.2:40886) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.737383 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1001 at 10.0.1.2:40886) Running State Change CS_EXCHANGE_MEDIA
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.737383 [DEBUG] switch_core_state_machine.c:542 (sofia/internal/1001 at 10.0.1.2:40886) State EXCHANGE_MEDIA
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.737383 [DEBUG] mod_sofia.c:613 SOFIA EXCHANGE_MEDIA
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:30.757403 [DEBUG] switch_rtp.c:6654 Correct audio ip/port confirmed.
send 1094 bytes to udp/[66.166.9.194]:39164 at 13:47:31.244642:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 10.0.1.12:40886;branch=z9hG4bK1578402832;rport=39164;received=66.166.9.194
   From: <sip:1000 at 216.209.211.222>;tag=243846933
   To: <sip:1001 at 216.209.211.222>;tag=e1FF6S7cgamNg
   Call-ID: 930817686-40886-5 at BA.A.B.BC
   CSeq: 41 INVITE
   Contact: <sip:1001 at 216.209.211.222:5060;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Require: timer
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Session-Expires: 120;refresher=uac
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 222
   P-Asserted-Identity: "Outbound Call" <sip:1001 at 216.209.211.222>
   
   v=0
   o=FreeSWITCH 1455720482 1455720483 IN IP4 216.209.211.222
   s=FreeSWITCH
   c=IN IP4 216.209.211.222
   t=0 0
   m=audio 25166 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
recv 764 bytes from udp/[66.166.9.194]:39164 at 13:47:31.321742:
   ------------------------------------------------------------------------
   ACK sip:1001 at 216.209.211.222:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 10.0.1.12:40886;branch=z9hG4bK145924562;rport
   From: <sip:1000 at 216.209.211.222>;tag=243846933
   To: <sip:1001 at 216.209.211.222>;tag=e1FF6S7cgamNg
   Call-ID: 930817686-40886-5 at BA.A.B.BC
   CSeq: 41 ACK
   Contact: <sip:1000 at 10.0.1.12:40886>
   Proxy-Authorization: Digest username="1000", realm="216.209.211.222", nonce="696ccc87-c214-4a8a-870f-7e3d69ca016d", uri="sip:1001 at 216.209.211.222", response="73ea6df6242e1f2e5cc6e4d6bad9be3e", algorithm=MD5, cnonce="15059615", qop=auth, nc=00000006
   Max-Forwards: 70
   Supported: replaces, path, timer, eventlist
   User-Agent: Grandstream Wave/IOS 1.0.19
   Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
   Content-Length: 0
   
   ------------------------------------------------------------------------
recv 764 bytes from udp/[66.166.9.194]:39164 at 13:47:31.327208:
   ------------------------------------------------------------------------
   ACK sip:1001 at 216.209.211.222:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 10.0.1.12:40886;branch=z9hG4bK145924562;rport
   From: <sip:1000 at 216.209.211.222>;tag=243846933
   To: <sip:1001 at 216.209.211.222>;tag=e1FF6S7cgamNg
   Call-ID: 930817686-40886-5 at BA.A.B.BC
   CSeq: 41 ACK
   Contact: <sip:1000 at 10.0.1.12:40886>
   Proxy-Authorization: Digest username="1000", realm="216.209.211.222", nonce="696ccc87-c214-4a8a-870f-7e3d69ca016d", uri="sip:1001 at 216.209.211.222", response="73ea6df6242e1f2e5cc6e4d6bad9be3e", algorithm=MD5, cnonce="15059615", qop=auth, nc=00000006
   Max-Forwards: 70
   Supported: replaces, path, timer, eventlist
   User-Agent: Grandstream Wave/IOS 1.0.19
   Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
   Content-Length: 0
   
   ------------------------------------------------------------------------
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:31.317425 [DEBUG] sofia.c:6760 Channel sofia/internal/1000 at 216.209.211.222 entering state [ready][200]
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:31.517381 [INFO] switch_rtp.c:6616 Auto Changing audio port from 10.0.1.2:35544 to 66.166.9.194:46630
recv 519 bytes from udp/[66.166.9.194]:39164 at 13:47:37.588043:
   ------------------------------------------------------------------------
   BYE sip:1001 at 216.209.211.222:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 10.0.1.12:40886;branch=z9hG4bK1092657280;rport
   From: <sip:1000 at 216.209.211.222>;tag=243846933
   To: <sip:1001 at 216.209.211.222>;tag=e1FF6S7cgamNg
   Call-ID: 930817686-40886-5 at BA.A.B.BC
   CSeq: 42 BYE
   Contact: <sip:1000 at 10.0.1.12:40886>
   Max-Forwards: 70
   Supported: replaces, path, timer, eventlist
   User-Agent: Grandstream Wave/IOS 1.0.19
   Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
   Content-Length: 0
   
   ------------------------------------------------------------------------
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [NOTICE] sofia.c:952 Hangup sofia/internal/1000 at 216.209.211.222 [CS_EXECUTE] [NORMAL_CLEARING]
send 466 bytes to udp/[66.166.9.194]:39164 at 13:47:37.598615:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 10.0.1.12:40886;branch=z9hG4bK1092657280;rport=39164;received=66.166.9.194
   From: <sip:1000 at 216.209.211.222>;tag=243846933
   To: <sip:1001 at 216.209.211.222>;tag=e1FF6S7cgamNg
   Call-ID: 930817686-40886-5 at BA.A.B.BC
   CSeq: 42 BYE
   User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Content-Length: 0
   
   ------------------------------------------------------------------------
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_ivr_bridge.c:778 BRIDGE THREAD DONE [sofia/internal/1000 at 216.209.211.222]
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_ivr_bridge.c:699 sofia/internal/1000 at 216.209.211.222 ending bridge by request from write function
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_ivr_bridge.c:778 BRIDGE THREAD DONE [sofia/internal/1001 at 10.0.1.2:40886]
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [NOTICE] switch_ivr_bridge.c:881 Hangup sofia/internal/1001 at 10.0.1.2:40886 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:542 (sofia/internal/1001 at 10.0.1.2:40886) State EXCHANGE_MEDIA going to sleep
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1001 at 10.0.1.2:40886) Running State Change CS_HANGUP
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:739 (sofia/internal/1001 at 10.0.1.2:40886) Callstate Change ACTIVE -> HANGUP
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_ivr_bridge.c:1689 sofia/internal/1001 at 10.0.1.2:40886 skip receive message [UNBRIDGE] (channel is hungup already)
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_ivr_bridge.c:1692 sofia/internal/1000 at 216.209.211.222 skip receive message [UNBRIDGE] (channel is hungup already)
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/1001 at 10.0.1.2:40886) State HANGUP
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] mod_sofia.c:425 sofia/internal/1001 at 10.0.1.2:40886 Overriding SIP cause 480 with 200 from the other leg
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] mod_sofia.c:431 Channel sofia/internal/1001 at 10.0.1.2:40886 hanging up, cause: NORMAL_CLEARING
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_session.c:2796 sofia/internal/1000 at 216.209.211.222 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:539 (sofia/internal/1000 at 216.209.211.222) State EXECUTE going to sleep
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1000 at 216.209.211.222) Running State Change CS_HANGUP
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:739 (sofia/internal/1000 at 216.209.211.222) Callstate Change ACTIVE -> HANGUP
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/1000 at 216.209.211.222) State HANGUP
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] mod_sofia.c:431 Channel sofia/internal/1000 at 216.209.211.222 hanging up, cause: NORMAL_CLEARING
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] mod_sofia.c:484 Sending BYE to sofia/internal/1001 at 10.0.1.2:40886
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1001 at 10.0.1.2:40886 Standard HANGUP, cause: NORMAL_CLEARING
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/1001 at 10.0.1.2:40886) State HANGUP going to sleep
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:508 (sofia/internal/1001 at 10.0.1.2:40886) State Change CS_HANGUP -> CS_REPORTING
send 559 bytes to udp/[66.166.9.194]:41958 at 13:47:37.609113:
   ------------------------------------------------------------------------
   BYE sip:1001 at 10.0.1.2:40886 SIP/2.0
   Via: SIP/2.0/UDP 216.209.211.222;rport;branch=z9hG4bK5tgvXv1evDy0c
   Max-Forwards: 70
   From: "Extension 1000" <sip:1000 at 216.209.211.222>;tag=Fa977mrgDKa8B
   To: <sip:1001 at 10.0.1.2:40886>;tag=2072085022
   Call-ID: e124e149-5062-1234-9bb4-c6827b410b7a
   CSeq: 87536760 BYE15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1001 at 10.0.1.2:40886) Running State Change CS_REPORTING

   User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Reason: Q.850;cause=16;text="NORMAL_CLEARING"
   Content-Length: 0
   
   ------------------------------------------------------------------------
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/1001 at 10.0.1.2:40886) State REPORTING
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:104 sofia/internal/1001 at 10.0.1.2:40886 Standard REPORTING, cause: NORMAL_CLEARING
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/1001 at 10.0.1.2:40886) State REPORTING going to sleep
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/1001 at 10.0.1.2:40886) State Change CS_REPORTING -> CS_DESTROY
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_session.c:1646 Session 7 (sofia/internal/1001 at 10.0.1.2:40886) Locked, Waiting on external entities
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [NOTICE] switch_core_session.c:1664 Session 7 (sofia/internal/1001 at 10.0.1.2:40886) Ended
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [NOTICE] switch_core_session.c:1668 Close Channel sofia/internal/1001 at 10.0.1.2:40886 [CS_DESTROY]
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1000 at 216.209.211.222 Standard HANGUP, cause: NORMAL_CLEARING
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/1000 at 216.209.211.222) State HANGUP going to sleep
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:508 (sofia/internal/1000 at 216.209.211.222) State Change CS_HANGUP -> CS_REPORTING
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:630 (sofia/internal/1001 at 10.0.1.2:40886) Running State Change CS_DESTROY
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/1001 at 10.0.1.2:40886) State DESTROY
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] mod_sofia.c:341 sofia/internal/1001 at 10.0.1.2:40886 SOFIA DESTROY
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1000 at 216.209.211.222) Running State Change CS_REPORTING
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:111 sofia/internal/1001 at 10.0.1.2:40886 Standard DESTROY
15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/1001 at 10.0.1.2:40886) State DESTROY going to sleep
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/1000 at 216.209.211.222) State REPORTING
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:104 sofia/internal/1000 at 216.209.211.222 Standard REPORTING, cause: NORMAL_CLEARING
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/1000 at 216.209.211.222) State REPORTING going to sleep
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/1000 at 216.209.211.222) State Change CS_REPORTING -> CS_DESTROY
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_session.c:1646 Session 6 (sofia/internal/1000 at 216.209.211.222) Locked, Waiting on external entities
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [NOTICE] switch_core_session.c:1664 Session 6 (sofia/internal/1000 at 216.209.211.222) Ended
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [NOTICE] switch_core_session.c:1668 Close Channel sofia/internal/1000 at 216.209.211.222 [CS_DESTROY]
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:630 (sofia/internal/1000 at 216.209.211.222) Running State Change CS_DESTROY
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/1000 at 216.209.211.222) State DESTROY
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] mod_sofia.c:341 sofia/internal/1000 at 216.209.211.222 SOFIA DESTROY
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:111 sofia/internal/1000 at 216.209.211.222 Standard DESTROY
4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/1000 at 216.209.211.222) State DESTROY going to sleep
recv 501 bytes from udp/[66.166.9.194]:41958 at 13:47:37.651003:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 216.209.211.222;rport=5060;branch=z9hG4bK5tgvXv1evDy0c
   From: "Extension 1000" <sip:1000 at 216.209.211.222>;tag=Fa977mrgDKa8B
   To: <sip:1001 at 10.0.1.2:40886>;tag=2072085022
   Call-ID: e124e149-5062-1234-9bb4-c6827b410b7a
   CSeq: 87536760 BYE
   Contact: <sip:1001 at 10.0.1.2:40886>
   Supported: replaces, path, timer, eventlist
   User-Agent: Grandstream Wave/IOS 1.0.19
   Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
   Content-Length: 0
   
   ------------------------------------------------------------------------
2016-02-17 13:47:41.537380 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/1000.csv
2016-02-17 13:47:41.537380 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/Master.csv

From Alexander.Haugg at c4b.de  Thu Feb 18 11:29:38 2016
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Thu, 18 Feb 2016 08:29:38 +0000
Subject: [Freeswitch-users] Web RTC -> No Candidate found
Message-ID: <B60DA91F5EA0E24A96E73CDE43CC9EC8C0059F58@vc4bex2.corp.c4b.de>

Hi,

I have the problem, that the freeswitch don't accept the network internal candidate.

The local IP address of the fs is 192.168.241.5
The local IP address of the webrtc Client is 192.168.241.2

In the acl.conf.xml is set:
<list name="localnet.auto" default="allow">
                <node type="allow" cidr="192.168.241.0/24"/>
</list>

In the sofia profile is set:
<param name="local-network-acl" value="localnet.auto"/>
<param name="apply-inbound-acl" value="localnet.auto"/>

The following SDP is send by the client:
v=0
o=- 222075295642546336 815544600 IN IP4 127.0.0.1
s=IceLink
t=0 0
m=audio 30450 RTP/SAVPF 0 8
c=IN IP4 192.168.62.1
a=rtcp:30450 IN IP4 192.168.62.1
a=ice-ufrag:532479e3
a=ice-pwd:4b066f2e64633d218085fc39c49a443c
a=sendrecv
a=rtcp-mux
a=fingerprint:sha-256 FD:DE:DB:44:C8:66:10:B0:DF:3A:D5:17:4D:89:0E:1A:39:0C:99:D3:AE:BE:1F:F2:D7:78:2E:8E:A6:6B:F1:36
a=setup:actpass
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ssrc:967296980 cname:c2088983
a=candidate:954dc3ed263236386e8615555070c83dg 1 udp 2130706331 192.168.241.2 44376 typ host
a=candidate:954dc3ed263236386e8615555070c83dg 2 udp 2130706331 192.168.241.2 44376 typ host

The output on the fs_cli:
2016-02-18 07:36:33.568868 [DEBUG] switch_channel.c:3759 (sofia/H3KSip/170) Callstate Change RINGING -> ACTIVE
2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4208 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4208 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:2898 Set Codec sofia/H3KSip_B2Bua/170 at 192.168.241.2:5058 PCMU/8000 20 ms 160 samples 64000 bits 1 channels
2016-02-18 07:36:33.588870 [DEBUG] switch_core_codec.c:111 sofia/H3KSip_B2Bua/170 at 192.168.241.2:5058 Original read codec set to PCMU:0
2016-02-18 07:36:33.588870 [WARNING] switch_core_media.c:3266 NO candidate ACL defined, Defaulting to wan.auto
2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3296 Save audio Candidate cid: 1 proto: udp type: host addr: 192.168.241.2:44376
2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3296 Save audio Candidate cid: 2 proto: udp type: host addr: 192.168.241.2:443760
2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3336 Searching for rtp candidate.
2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3336 Searching for rtcp candidate.
2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3380 sofia/H3KSip_B2Bua/170 at 192.168.241.2:5058 no suitable candidates found.
2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4480 No 2833 in SDP.  Disable 2833 dtmf and switch to INFO
2016-02-18 07:36:33.588870 [NOTICE] switch_channel.c:3798 Hangup sofia/H3KSip_B2Bua/170 at 192.168.241.2:5058 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION

Thanks a lot!

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From brian at freeswitch.org  Thu Feb 18 17:31:25 2016
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Feb 2016 08:31:25 -0600
Subject: [Freeswitch-users] Web RTC -> No Candidate found
In-Reply-To: <B60DA91F5EA0E24A96E73CDE43CC9EC8C0059F58@vc4bex2.corp.c4b.de>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC8C0059F58@vc4bex2.corp.c4b.de>
Message-ID: <CAEJMVkAn26kvUtFMtq+U+kPsTEjR-3FHxD1mFADXUqsZuw=gww@mail.gmail.com>

Try

apply-candidate-acl

On Thu, Feb 18, 2016 at 2:29 AM, Alexander Haugg <Alexander.Haugg at c4b.de>
wrote:

> Hi,
>
>
>
> I have the problem, that the freeswitch don?t accept the network internal
> candidate.
>
>
>
> The local IP address of the fs is 192.168.241.5
>
> The local IP address of the webrtc Client is 192.168.241.2
>
>
>
> In the acl.conf.xml is set:
>
> <list name="localnet.auto" default="allow">
>
>                 <node type="allow" cidr="192.168.241.0/24"/>
>
> </list>
>
>
>
> In the sofia profile is set:
>
> <param name="local-network-acl" value="localnet.auto"/>
>
> <param name="apply-inbound-acl" value="localnet.auto"/>
>
>
>
> The following SDP is send by the client:
>
> v=0
>
> o=- 222075295642546336 815544600 IN IP4 127.0.0.1
>
> s=IceLink
>
> t=0 0
>
> m=audio 30450 RTP/SAVPF 0 8
>
> c=IN IP4 192.168.62.1
>
> a=rtcp:30450 IN IP4 192.168.62.1
>
> a=ice-ufrag:532479e3
>
> a=ice-pwd:4b066f2e64633d218085fc39c49a443c
>
> a=sendrecv
>
> a=rtcp-mux
>
> a=fingerprint:sha-256
> FD:DE:DB:44:C8:66:10:B0:DF:3A:D5:17:4D:89:0E:1A:39:0C:99:D3:AE:BE:1F:F2:D7:78:2E:8E:A6:6B:F1:36
>
> a=setup:actpass
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:8 PCMA/8000
>
> a=ssrc:967296980 cname:c2088983
>
> a=candidate:954dc3ed263236386e8615555070c83dg 1 udp 2130706331
> 192.168.241.2 44376 typ host
>
> a=candidate:954dc3ed263236386e8615555070c83dg 2 udp 2130706331
> 192.168.241.2 44376 typ host
>
>
>
> The output on the fs_cli:
>
> 2016-02-18 07:36:33.568868 [DEBUG] switch_channel.c:3759
> (sofia/H3KSip/170) Callstate Change RINGING -> ACTIVE
>
> 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec
> Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
>
> 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4208 Audio Codec
> Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
>
> 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec
> Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>
> 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec
> Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
>
> 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec
> Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>
> 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4208 Audio Codec
> Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
>
> 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:2898 Set Codec
> sofia/H3KSip_B2Bua/170 at 192.168.241.2:5058 PCMU/8000 20 ms 160 samples
> 64000 bits 1 channels
>
> 2016-02-18 07:36:33.588870 [DEBUG] switch_core_codec.c:111
> sofia/H3KSip_B2Bua/170 at 192.168.241.2:5058 Original read codec set to
> PCMU:0
>
> 2016-02-18 07:36:33.588870 [WARNING] switch_core_media.c:3266 NO candidate
> ACL defined, Defaulting to wan.auto
>
> 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3296 Save audio
> Candidate cid: 1 proto: udp type: host addr: 192.168.241.2:44376
>
> 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3296 Save audio
> Candidate cid: 2 proto: udp type: host addr: 192.168.241.2:443760
>
> 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3336 Searching for
> rtp candidate.
>
> 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3336 Searching for
> rtcp candidate.
>
> 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3380
> sofia/H3KSip_B2Bua/170 at 192.168.241.2:5058 no suitable candidates found.
>
> 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4480 No 2833 in
> SDP.  Disable 2833 dtmf and switch to INFO
>
> 2016-02-18 07:36:33.588870 [NOTICE] switch_channel.c:3798 Hangup
> sofia/H3KSip_B2Bua/170 at 192.168.241.2:5058 [CS_EXECUTE]
> [INCOMPATIBLE_DESTINATION
>
>
>
> 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
>



-- 

*Brian West*
brian at freeswitch.org


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

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 mike at jerris.com  Thu Feb 18 19:28:02 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 18 Feb 2016 11:28:02 -0500
Subject: [Freeswitch-users] new version of vpx needed for those tracking
	master.
Message-ID: <6F83920F-35F5-4B15-B9CC-AC5C4648C7DA@jerris.com>

Please note, we have updated code in master that will now require the newer version of libvpx (and anything such as libav that is linked against vpx will also need to be updated) in order to use video features.  The tarball for this newer version is located at http://files.freeswitch.org/downloads/libs/libvpx2-1.5.0.tar.gz .  If you are using our debian packages for dependencies, these packages are already available in the unstable repo.




From vagarwal at vertical.com  Thu Feb 18 21:49:45 2016
From: vagarwal at vertical.com (Varsha Agarwal)
Date: Thu, 18 Feb 2016 18:49:45 +0000
Subject: [Freeswitch-users] uuid_simplify or uuid_deflect in mod_callCenter
Message-ID: <04AB0A185A23864CA9255FD38901F65D023374A69A@SCEX1.vertical.com>

Hi,

Can I use uuid_simplify or uuid_deflect in CallCenter config to route agent calls? I want to send REFER with Replace for Agent transfers as our agents are registered to a different PBX and FreeSwitch is just a gateway. REFER will remove FreeSwitch from the call path. If yes, can I get an example, I tried using it in CallCenter config but I get error cannot locate uuid_simplify or cannot locate channel uuid_deflect.

Thanks,
Varsha
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From mafra.fabio at gmail.com  Thu Feb 18 22:02:12 2016
From: mafra.fabio at gmail.com (Fabio Mafra)
Date: Thu, 18 Feb 2016 17:02:12 -0200
Subject: [Freeswitch-users] Persistent caller id on the download links ...
Message-ID: <CAPRu2Dd9=nvGX=MgooxC7jEDVUijQqzFN4hHZNMP2b3wcNg2bw@mail.gmail.com>

I am a new user FreeSWITCH.
I found a flaw in the system right after installation to make a call
handover between 3 SIP extensions. Example:
Extension 1 Extension 2 calls.
2 extension meets and press the "Hold" button.
Extension 2 press the "transfer" button, and dials the number of the
extension 3.
Extension 2 and Extension 3 communicate.
Extension 2 press the "transfer" button again to complete the transfer of
extension 1 to extension 3.
Pressing the "transfer" the extension 2 comes out link.
Now Extension 1 and Extension 3 are communicating connection with the
extension 2.
But in Extension 1 and Extension Display 3 appear as if they were
communicating with Extension 2 (no longer part of the link).
After transfer to the Station 1 does not have any indication be
communicating with the Station 3?
After the transfer of the branch line 3 does not have any indication be
communicating with Extension 1?
How can I fix this problem?
My FreeSWITCH is installed on a Debian machine 8-Jessie at: / usr / local /
freeswitch.
I use the internal settings of extensions 1000.xml to 1019.xml.
Thank you for your attention right now!
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From mario_fs at mgtech.com  Thu Feb 18 22:56:20 2016
From: mario_fs at mgtech.com (Mario G)
Date: Thu, 18 Feb 2016 11:56:20 -0800
Subject: [Freeswitch-users] new version of vpx needed for those tracking
	master.
In-Reply-To: <6F83920F-35F5-4B15-B9CC-AC5C4648C7DA@jerris.com>
References: <6F83920F-35F5-4B15-B9CC-AC5C4648C7DA@jerris.com>
Message-ID: <F0579136-F83E-4EAD-B7D7-DB552297D9E4@mgtech.com>

FS build os now broken on OS X, I noticed that home-brew did not update libvpx. Assume it needs to be updated too?


> On Feb 18, 2016, at 8:28 AM, Michael Jerris <mike at jerris.com> wrote:
> 
> Please note, we have updated code in master that will now require the newer version of libvpx (and anything such as libav that is linked against vpx will also need to be updated) in order to use video features.  The tarball for this newer version is located at http://files.freeswitch.org/downloads/libs/libvpx2-1.5.0.tar.gz .  If you are using our debian packages for dependencies, these packages are already available in the unstable repo.
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Thu Feb 18 23:07:38 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 18 Feb 2016 15:07:38 -0500
Subject: [Freeswitch-users] new version of vpx needed for those tracking
	master.
In-Reply-To: <F0579136-F83E-4EAD-B7D7-DB552297D9E4@mgtech.com>
References: <6F83920F-35F5-4B15-B9CC-AC5C4648C7DA@jerris.com>
	<F0579136-F83E-4EAD-B7D7-DB552297D9E4@mgtech.com>
Message-ID: <E6CC8D5A-CB5A-4BA4-9A59-2E96233A7F53@jerris.com>

I have not updated it yet.. will need to address that soon.

> On Feb 18, 2016, at 2:56 PM, Mario G <mario_fs at mgtech.com> wrote:
> 
> FS build os now broken on OS X, I noticed that home-brew did not update libvpx. Assume it needs to be updated too?
> 
> 
>> On Feb 18, 2016, at 8:28 AM, Michael Jerris <mike at jerris.com> wrote:
>> 
>> Please note, we have updated code in master that will now require the newer version of libvpx (and anything such as libav that is linked against vpx will also need to be updated) in order to use video features.  The tarball for this newer version is located at http://files.freeswitch.org/downloads/libs/libvpx2-1.5.0.tar.gz .  If you are using our debian packages for dependencies, these packages are already available in the unstable repo.
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: 
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mario_fs at mgtech.com  Fri Feb 19 01:27:46 2016
From: mario_fs at mgtech.com (Mario G)
Date: Thu, 18 Feb 2016 14:27:46 -0800
Subject: [Freeswitch-users] new version of vpx needed for those tracking
	master.
In-Reply-To: <E6CC8D5A-CB5A-4BA4-9A59-2E96233A7F53@jerris.com>
References: <6F83920F-35F5-4B15-B9CC-AC5C4648C7DA@jerris.com>
	<F0579136-F83E-4EAD-B7D7-DB552297D9E4@mgtech.com>
	<E6CC8D5A-CB5A-4BA4-9A59-2E96233A7F53@jerris.com>
Message-ID: <15C74368-4311-40B8-A30C-1244A9C4917B@mgtech.com>

OK thank you, if you let me know I?ll test the 3 releases afterwards.
Mario G

> On Feb 18, 2016, at 12:07 PM, Michael Jerris <mike at jerris.com> wrote:
> 
> I have not updated it yet.. will need to address that soon.
> 
>> On Feb 18, 2016, at 2:56 PM, Mario G <mario_fs at mgtech.com> wrote:
>> 
>> FS build os now broken on OS X, I noticed that home-brew did not update libvpx. Assume it needs to be updated too?
>> 
>> 
>>> On Feb 18, 2016, at 8:28 AM, Michael Jerris <mike at jerris.com> wrote:
>>> 
>>> Please note, we have updated code in master that will now require the newer version of libvpx (and anything such as libav that is linked against vpx will also need to be updated) in order to use video features.  The tarball for this newer version is located at http://files.freeswitch.org/downloads/libs/libvpx2-1.5.0.tar.gz .  If you are using our debian packages for dependencies, these packages are already available in the unstable repo.
>>> 
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services: 
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: 
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 gascagonzalo at gmail.com  Fri Feb 19 07:59:51 2016
From: gascagonzalo at gmail.com (Gonzalo Gasca Meza)
Date: Thu, 18 Feb 2016 20:59:51 -0800
Subject: [Freeswitch-users] Firefox vs Chrome in Verto Communicator
	(self-signed certificates)
Message-ID: <CAL+DTgKjhS5=35MdwM+Gtyn4UZquZpbO_-DVobd6=SJJOUuhBA@mail.gmail.com>

I compiled FS 1.6 from source and using Verto Communicator with *Firefox*
shows:
"Waiting for server reconnection."
Firefox 44.0.2

   - MacOS system 10.10.4.
   - Windows 7



-*Works fine with Chrome* (I tested Chrome successfully to confirm there is
no network issues).
*-Same error in 2 different networks.*

It works only after I enter in Browser manually:
https://<IP>:8082 and accept the certificate

I'm using self-signed certificates. Is this Browser related or Application
related?
Looks like is Browser related, need confirmation.

*Console logs:*
http://pastebin.com/XSJA9dZf


*References:*
https://bugzilla.mozilla.org/show_bug.cgi?id=594502
https://freeswitch.org/jira/browse/FS-8805



[image: Inline image 1]
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From gmaruzz at gmail.com  Fri Feb 19 09:33:33 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 19 Feb 2016 07:33:33 +0100
Subject: [Freeswitch-users] Firefox vs Chrome in Verto Communicator
 (self-signed certificates)
In-Reply-To: <CAL+DTgKjhS5=35MdwM+Gtyn4UZquZpbO_-DVobd6=SJJOUuhBA@mail.gmail.com>
References: <CAL+DTgKjhS5=35MdwM+Gtyn4UZquZpbO_-DVobd6=SJJOUuhBA@mail.gmail.com>
Message-ID: <CALXCt0rCeY26Defm+WiMy_Tkc+oXe0C8C+d-JhSatYMGNcaU4g@mail.gmail.com>

As very clearly put in all documentation, self signed certificates are NOT
supported (they may work if you really know how to do).

Use real certificates.

Google "letsencrypt" and use their utility to get real certificates
completely for free.

-giovanni
Il 19/Feb/2016 06:01, "Gonzalo Gasca Meza" <gascagonzalo at gmail.com> ha
scritto:

> I compiled FS 1.6 from source and using Verto Communicator with *Firefox*
> shows:
> "Waiting for server reconnection."
> Firefox 44.0.2
>
>    - MacOS system 10.10.4.
>    - Windows 7
>
>
>
> -*Works fine with Chrome* (I tested Chrome successfully to confirm there
> is no network issues).
> *-Same error in 2 different networks.*
>
> It works only after I enter in Browser manually:
> https://<IP>:8082 and accept the certificate
>
> I'm using self-signed certificates. Is this Browser related or Application
> related?
> Looks like is Browser related, need confirmation.
>
> *Console logs:*
> http://pastebin.com/XSJA9dZf
>
>
> *References:*
> https://bugzilla.mozilla.org/show_bug.cgi?id=594502
> https://freeswitch.org/jira/browse/FS-8805
>
>
>
> [image: Inline image 1]
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 gascagonzalo at gmail.com  Fri Feb 19 09:43:30 2016
From: gascagonzalo at gmail.com (Gonzalo Gasca Meza)
Date: Thu, 18 Feb 2016 22:43:30 -0800
Subject: [Freeswitch-users] Firefox vs Chrome in Verto Communicator
 (self-signed certificates)
In-Reply-To: <CALXCt0rCeY26Defm+WiMy_Tkc+oXe0C8C+d-JhSatYMGNcaU4g@mail.gmail.com>
References: <CAL+DTgKjhS5=35MdwM+Gtyn4UZquZpbO_-DVobd6=SJJOUuhBA@mail.gmail.com>
	<CALXCt0rCeY26Defm+WiMy_Tkc+oXe0C8C+d-JhSatYMGNcaU4g@mail.gmail.com>
Message-ID: <CAL+DTgKND0crxJVG_07=5hUpEGapMOeivn7D_djMxFijCk9Yxg@mail.gmail.com>

Thanks Giovanni

On Thu, Feb 18, 2016 at 10:33 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> As very clearly put in all documentation, self signed certificates are NOT
> supported (they may work if you really know how to do).
>
> Use real certificates.
>
> Google "letsencrypt" and use their utility to get real certificates
> completely for free.
>
> -giovanni
> Il 19/Feb/2016 06:01, "Gonzalo Gasca Meza" <gascagonzalo at gmail.com> ha
> scritto:
>
>> I compiled FS 1.6 from source and using Verto Communicator with *Firefox*
>> shows:
>> "Waiting for server reconnection."
>> Firefox 44.0.2
>>
>>    - MacOS system 10.10.4.
>>    - Windows 7
>>
>>
>>
>> -*Works fine with Chrome* (I tested Chrome successfully to confirm there
>> is no network issues).
>> *-Same error in 2 different networks.*
>>
>> It works only after I enter in Browser manually:
>> https://<IP>:8082 and accept the certificate
>>
>> I'm using self-signed certificates. Is this Browser related or
>> Application related?
>> Looks like is Browser related, need confirmation.
>>
>> *Console logs:*
>> http://pastebin.com/XSJA9dZf
>>
>>
>> *References:*
>> https://bugzilla.mozilla.org/show_bug.cgi?id=594502
>> https://freeswitch.org/jira/browse/FS-8805
>>
>>
>>
>> [image: Inline image 1]
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Fri Feb 19 13:40:09 2016
From: aqsyounas at gmail.com (Aqs Younas)
Date: Fri, 19 Feb 2016 15:40:09 +0500
Subject: [Freeswitch-users] [ERR] mod_avmd.c:226 Couldn't register subclass
	avmd::beep!
Message-ID: <CAGVOjVXggKeDX2BNT7RaFxLiT_w2YdVUS2kBpZJgUX+cJL+7Ng@mail.gmail.com>

Hi.

I am able to compile mod_avmd without any error. But when i try to load it
i see this error in console.

freeswitch at internal> load mod_avmd
+OK Reloading XML
-ERR [module load file routine returned an error]

2016-02-19 05:27:31.644461 [INFO] mod_enum.c:880 ENUM Reloaded
2016-02-19 05:27:31.644461 [ERR] mod_avmd.c:226 Couldn't register subclass
avmd::beep!
2016-02-19 05:27:31.644461 [CRIT] switch_loadable_module.c:1447 Error
Loading module /usr/local/freeswitch/mod/mod_avmd.so
**Module load routine returned an error**


These are the lines I see in mod_avmd.c but unable to understand.


        if (switch_event_reserve_subclass(AVMD_EVENT_BEEP) !=
SWITCH_STATUS_SUCCESS) {
                switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR,
"Couldn't register subclass %s!\n", AVMD_EVENT_BEEP);
                return SWITCH_STATUS_TERM;
        }


Any pointer is much appreciated.

Best Regards.

This email has been sent from a virus-free computer protected by Avast.
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From vma at 440hz.fr  Fri Feb 19 14:33:10 2016
From: vma at 440hz.fr (Vallimamod Abdullah)
Date: Fri, 19 Feb 2016 12:33:10 +0100
Subject: [Freeswitch-users] [ERR] mod_avmd.c:226 Couldn't register
	subclass avmd::beep!
In-Reply-To: <CAGVOjVXggKeDX2BNT7RaFxLiT_w2YdVUS2kBpZJgUX+cJL+7Ng@mail.gmail.com>
References: <CAGVOjVXggKeDX2BNT7RaFxLiT_w2YdVUS2kBpZJgUX+cJL+7Ng@mail.gmail.com>
Message-ID: <22E5B8B0-1C62-4BD1-9B94-32ECF3D5E0A5@440hz.fr>

Hi,

Looks like you are trying to load a module that is already loaded. Try a reload. Also, sometimes the event subclass doesn?t unregister properly when you do a reload so you need to restart freeswitch when you get this error.

Best Regards,
Vallimamod
.


> On 19 Feb 2016, at 11:40, Aqs Younas <aqsyounas at gmail.com> wrote:
> 
> Hi.
> 
> I am able to compile mod_avmd without any error. But when i try to load it i see this error in console. 
> 
> freeswitch at internal> load mod_avmd
> +OK Reloading XML
> -ERR [module load file routine returned an error]
> 
> 2016-02-19 05:27:31.644461 [INFO] mod_enum.c:880 ENUM Reloaded
> 2016-02-19 05:27:31.644461 [ERR] mod_avmd.c:226 Couldn't register subclass avmd::beep!
> 2016-02-19 05:27:31.644461 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_avmd.so
> **Module load routine returned an error**
> 
> 
> These are the lines I see in mod_avmd.c but unable to understand. 
> 
> 
>         if (switch_event_reserve_subclass(AVMD_EVENT_BEEP) != SWITCH_STATUS_SUCCESS) {
>                 switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Couldn't register subclass %s!\n", AVMD_EVENT_BEEP);
>                 return SWITCH_STATUS_TERM;
>         }
> 
> 
> Any pointer is much appreciated. 
> 
> Best Regards.

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From aqsyounas at gmail.com  Fri Feb 19 14:56:55 2016
From: aqsyounas at gmail.com (Aqs Younas)
Date: Fri, 19 Feb 2016 16:56:55 +0500
Subject: [Freeswitch-users] [ERR] mod_avmd.c:226 Couldn't register
 subclass avmd::beep!
In-Reply-To: <22E5B8B0-1C62-4BD1-9B94-32ECF3D5E0A5@440hz.fr>
References: <CAGVOjVXggKeDX2BNT7RaFxLiT_w2YdVUS2kBpZJgUX+cJL+7Ng@mail.gmail.com>
	<22E5B8B0-1C62-4BD1-9B94-32ECF3D5E0A5@440hz.fr>
Message-ID: <CAGVOjVVrD2BqwCmTc5-ab=QFbbBRf60so0gmsa14=3X55uN0EQ@mail.gmail.com>

Perfect. Thank you. !

This email has been sent from a virus-free computer protected by Avast.
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<#DDB4FAA8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

On 19 February 2016 at 16:33, Vallimamod Abdullah <vma at 440hz.fr> wrote:

> Hi,
>
> Looks like you are trying to load a module that is already loaded. Try a
> reload. Also, sometimes the event subclass doesn?t unregister properly when
> you do a reload so you need to restart freeswitch when you get this error.
>
> Best Regards,
> Vallimamod
> .
>
>
> On 19 Feb 2016, at 11:40, Aqs Younas <aqsyounas at gmail.com> wrote:
>
> Hi.
>
> I am able to compile mod_avmd without any error. But when i try to load it
> i see this error in console.
>
> freeswitch at internal> load mod_avmd
> +OK Reloading XML
> -ERR [module load file routine returned an error]
>
> 2016-02-19 05:27:31.644461 [INFO] mod_enum.c:880 ENUM Reloaded
> 2016-02-19 05:27:31.644461 [ERR] mod_avmd.c:226 Couldn't register subclass
> avmd::beep!
> 2016-02-19 05:27:31.644461 [CRIT] switch_loadable_module.c:1447 Error
> Loading module /usr/local/freeswitch/mod/mod_avmd.so
> **Module load routine returned an error**
>
>
> These are the lines I see in mod_avmd.c but unable to understand.
>
>
>         if (switch_event_reserve_subclass(AVMD_EVENT_BEEP) !=
> SWITCH_STATUS_SUCCESS) {
>                 switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR,
> "Couldn't register subclass %s!\n", AVMD_EVENT_BEEP);
>                 return SWITCH_STATUS_TERM;
>         }
>
>
> Any pointer is much appreciated.
>
> 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 brian at freeswitch.org  Fri Feb 19 15:44:19 2016
From: brian at freeswitch.org (Brian West)
Date: Fri, 19 Feb 2016 06:44:19 -0600
Subject: [Freeswitch-users] [ERR] mod_avmd.c:226 Couldn't register
 subclass avmd::beep!
In-Reply-To: <CAGVOjVXggKeDX2BNT7RaFxLiT_w2YdVUS2kBpZJgUX+cJL+7Ng@mail.gmail.com>
References: <CAGVOjVXggKeDX2BNT7RaFxLiT_w2YdVUS2kBpZJgUX+cJL+7Ng@mail.gmail.com>
Message-ID: <CAEJMVkAnCBO-7tSFG6L=dxLDUCYTSXbP9BCrAVMvavf4ucQs2A@mail.gmail.com>

[image: Inline image 1]


On Fri, Feb 19, 2016 at 4:40 AM, Aqs Younas <aqsyounas at gmail.com> wrote:

> Hi.
>
> I am able to compile mod_avmd without any error. But when i try to load it
> i see this error in console.
>
> freeswitch at internal> load mod_avmd
> +OK Reloading XML
> -ERR [module load file routine returned an error]
>
> 2016-02-19 05:27:31.644461 [INFO] mod_enum.c:880 ENUM Reloaded
> 2016-02-19 05:27:31.644461 [ERR] mod_avmd.c:226 Couldn't register subclass
> avmd::beep!
> 2016-02-19 05:27:31.644461 [CRIT] switch_loadable_module.c:1447 Error
> Loading module /usr/local/freeswitch/mod/mod_avmd.so
> **Module load routine returned an error**
>
>
> These are the lines I see in mod_avmd.c but unable to understand.
>
>
>         if (switch_event_reserve_subclass(AVMD_EVENT_BEEP) !=
> SWITCH_STATUS_SUCCESS) {
>                 switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR,
> "Couldn't register subclass %s!\n", AVMD_EVENT_BEEP);
>                 return SWITCH_STATUS_TERM;
>         }
>
>
> Any pointer is much appreciated.
>
> Best Regards.
>
> This email has been sent from a virus-free computer protected by Avast.
> www.avast.com <https://www.avast.com/sig-email>
> <#1121851626_DDB4FAA8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

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 aqsyounas at gmail.com  Fri Feb 19 15:59:53 2016
From: aqsyounas at gmail.com (Aqs Younas)
Date: Fri, 19 Feb 2016 17:59:53 +0500
Subject: [Freeswitch-users] [ERR] mod_avmd.c:226 Couldn't register
 subclass avmd::beep!
In-Reply-To: <CAEJMVkAnCBO-7tSFG6L=dxLDUCYTSXbP9BCrAVMvavf4ucQs2A@mail.gmail.com>
References: <CAGVOjVXggKeDX2BNT7RaFxLiT_w2YdVUS2kBpZJgUX+cJL+7Ng@mail.gmail.com>
	<CAEJMVkAnCBO-7tSFG6L=dxLDUCYTSXbP9BCrAVMvavf4ucQs2A@mail.gmail.com>
Message-ID: <CAGVOjVUJKyufW+s3KhB0B66zVvN=9y=E_03xYWOVQzS=5Het8w@mail.gmail.com>

I was using NewFies Dialer :)

On 19 February 2016 at 17:44, Brian West <brian at freeswitch.org> wrote:

> [image: Inline image 1]
>
>
> On Fri, Feb 19, 2016 at 4:40 AM, Aqs Younas <aqsyounas at gmail.com> wrote:
>
>> Hi.
>>
>> I am able to compile mod_avmd without any error. But when i try to load
>> it i see this error in console.
>>
>> freeswitch at internal> load mod_avmd
>> +OK Reloading XML
>> -ERR [module load file routine returned an error]
>>
>> 2016-02-19 05:27:31.644461 [INFO] mod_enum.c:880 ENUM Reloaded
>> 2016-02-19 05:27:31.644461 [ERR] mod_avmd.c:226 Couldn't register
>> subclass avmd::beep!
>> 2016-02-19 05:27:31.644461 [CRIT] switch_loadable_module.c:1447 Error
>> Loading module /usr/local/freeswitch/mod/mod_avmd.so
>> **Module load routine returned an error**
>>
>>
>> These are the lines I see in mod_avmd.c but unable to understand.
>>
>>
>>         if (switch_event_reserve_subclass(AVMD_EVENT_BEEP) !=
>> SWITCH_STATUS_SUCCESS) {
>>                 switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR,
>> "Couldn't register subclass %s!\n", AVMD_EVENT_BEEP);
>>                 return SWITCH_STATUS_TERM;
>>         }
>>
>>
>> Any pointer is much appreciated.
>>
>> Best Regards.
>>
>> This email has been sent from a virus-free computer protected by Avast.
>> www.avast.com <https://www.avast.com/sig-email>
>> <#-55043839_1121851626_DDB4FAA8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> 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 areski at gmail.com  Fri Feb 19 17:39:16 2016
From: areski at gmail.com (Areski)
Date: Fri, 19 Feb 2016 15:39:16 +0100
Subject: [Freeswitch-users] Live Monitoring FreeSWITCH with Grafana,
	InfluxDB & Telegraf
Message-ID: <CAMf7LBh2jjOUf0sbAMyzteCwQ0PrMyh8QLLe5X=27uCoEfVy8Q@mail.gmail.com>

Hi,

If you are interested how to build this Live monitoring system for
FreeSWITCH using Grafana & InfluxDB, we have written a post describing how:
http://www.newfies-dialer.org/live-monitoring-freeswitch-newfies-dialer-with-grafana-influxdb-telegraf/

I would love to hear how you do your own monitoring or how we could improve
this.

-- 
Kind regards,
/Areski

----
Arezqui Belaid, <areski at gmail.com>
Founder at Star2Billing (www.star2billing.com)

Tel: +34650784355
Twitter: http://twitter.com/areskib
LinkedIn: http://www.linkedin.com/in/areski
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From gmaruzz at gmail.com  Fri Feb 19 17:53:35 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 19 Feb 2016 15:53:35 +0100
Subject: [Freeswitch-users] Live Monitoring FreeSWITCH with Grafana,
 InfluxDB & Telegraf
In-Reply-To: <CAMf7LBh2jjOUf0sbAMyzteCwQ0PrMyh8QLLe5X=27uCoEfVy8Q@mail.gmail.com>
References: <CAMf7LBh2jjOUf0sbAMyzteCwQ0PrMyh8QLLe5X=27uCoEfVy8Q@mail.gmail.com>
Message-ID: <CALXCt0r6szEBcz3kHV1oAdPjBG7cOL59TV8TUcAQH6jLdLp_wg@mail.gmail.com>

Yay Areski!!!!!
Looking forward to read that beauty !
-giovanni
Il 19/Feb/2016 15:40, "Areski" <areski at gmail.com> ha scritto:

> Hi,
>
> If you are interested how to build this Live monitoring system for
> FreeSWITCH using Grafana & InfluxDB, we have written a post describing how:
>
> http://www.newfies-dialer.org/live-monitoring-freeswitch-newfies-dialer-with-grafana-influxdb-telegraf/
>
> I would love to hear how you do your own monitoring or how we could
> improve this.
>
> --
> Kind regards,
> /Areski
>
> ----
> Arezqui Belaid, <areski at gmail.com>
> Founder at Star2Billing (www.star2billing.com)
>
> Tel: +34650784355
> Twitter: http://twitter.com/areskib
> LinkedIn: http://www.linkedin.com/in/areski
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gmaruzz at gmail.com  Fri Feb 19 18:22:53 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 19 Feb 2016 16:22:53 +0100
Subject: [Freeswitch-users] Live Monitoring FreeSWITCH with Grafana,
 InfluxDB & Telegraf
In-Reply-To: <CAMf7LBh2jjOUf0sbAMyzteCwQ0PrMyh8QLLe5X=27uCoEfVy8Q@mail.gmail.com>
References: <CAMf7LBh2jjOUf0sbAMyzteCwQ0PrMyh8QLLe5X=27uCoEfVy8Q@mail.gmail.com>
Message-ID: <CALXCt0r-D9eo_u1CJuTrAa-SDj6Vy1A+qSvkGKoj7w7DryfAsA@mail.gmail.com>

Very nice post and solution! Looking forward to install and test it.

We use Homer (Voipcapture) + check_mk

-giovanni

On Fri, Feb 19, 2016 at 3:39 PM, Areski <areski at gmail.com> wrote:

> Hi,
>
> If you are interested how to build this Live monitoring system for
> FreeSWITCH using Grafana & InfluxDB, we have written a post describing how:
>
> http://www.newfies-dialer.org/live-monitoring-freeswitch-newfies-dialer-with-grafana-influxdb-telegraf/
>
> I would love to hear how you do your own monitoring or how we could
> improve this.
>
> --
> Kind regards,
> /Areski
>
> ----
> Arezqui Belaid, <areski at gmail.com>
> Founder at Star2Billing (www.star2billing.com)
>
> Tel: +34650784355
> Twitter: http://twitter.com/areskib
> LinkedIn: http://www.linkedin.com/in/areski
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 areski at gmail.com  Fri Feb 19 18:57:37 2016
From: areski at gmail.com (Areski)
Date: Fri, 19 Feb 2016 16:57:37 +0100
Subject: [Freeswitch-users] Live Monitoring FreeSWITCH with Grafana,
 InfluxDB & Telegraf
In-Reply-To: <CALXCt0r-D9eo_u1CJuTrAa-SDj6Vy1A+qSvkGKoj7w7DryfAsA@mail.gmail.com>
References: <CAMf7LBh2jjOUf0sbAMyzteCwQ0PrMyh8QLLe5X=27uCoEfVy8Q@mail.gmail.com>
	<CALXCt0r-D9eo_u1CJuTrAa-SDj6Vy1A+qSvkGKoj7w7DryfAsA@mail.gmail.com>
Message-ID: <CAMf7LBj3xYemKOvZQsr5Bxf-huVVqGBQwH7dNZq_uSFM=wUevw@mail.gmail.com>

Thanks for the feedback Giovanni, I will definitely look at Homer for
quality monitoring, didnt had time to play with it yet :)
What was really important for us now it's to provide a multitenant UI where
users could see their own metrics but also measure vital application
metrics like redis, postgresql, etc...

On Fri, Feb 19, 2016 at 4:22 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> Very nice post and solution! Looking forward to install and test it.
>
> We use Homer (Voipcapture) + check_mk
>
> -giovanni
>
> On Fri, Feb 19, 2016 at 3:39 PM, Areski <areski at gmail.com> wrote:
>
>> Hi,
>>
>> If you are interested how to build this Live monitoring system for
>> FreeSWITCH using Grafana & InfluxDB, we have written a post describing how:
>>
>> http://www.newfies-dialer.org/live-monitoring-freeswitch-newfies-dialer-with-grafana-influxdb-telegraf/
>>
>> I would love to hear how you do your own monitoring or how we could
>> improve this.
>>
>> --
>> Kind regards,
>> /Areski
>>
>> ----
>> Arezqui Belaid, <areski at gmail.com>
>> Founder at Star2Billing (www.star2billing.com)
>>
>> Tel: +34650784355
>> Twitter: http://twitter.com/areskib
>> LinkedIn: http://www.linkedin.com/in/areski
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>



-- 
Kind regards,
/Areski

----
Arezqui Belaid, <areski at gmail.com>
Founder at Star2Billing (www.star2billing.com)

Tel: +34650784355
Twitter: http://twitter.com/areskib
LinkedIn: http://www.linkedin.com/in/areski
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From martin.hoole at emailn.de  Fri Feb 19 19:31:46 2016
From: martin.hoole at emailn.de (martin.hoole at emailn.de)
Date: Fri, 19 Feb 2016 17:31:46 +0100
Subject: [Freeswitch-users] Selective Forwarding Unit (SFU)
Message-ID: <73dcf231ea663b1f127af8b41efcb108@mail.emailn.de>

Are there plans to implement support for Selective Forwarding Unit (SFU) for video or audio rather than mixing video together?
https://tools.ietf.org/html/draft-ietf-avtcore-rtp-topologies-update-01




From anthony.minessale at gmail.com  Fri Feb 19 19:40:28 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Feb 2016 10:40:28 -0600
Subject: [Freeswitch-users] Selective Forwarding Unit (SFU)
In-Reply-To: <73dcf231ea663b1f127af8b41efcb108@mail.emailn.de>
References: <73dcf231ea663b1f127af8b41efcb108@mail.emailn.de>
Message-ID: <CAKbxfG9nmuxC6hWJDDBTxcMoGt3bWRfBbucWeCp_6y1D=Bc_nA@mail.gmail.com>

Not without a massive demand for it from the user base and appropriate
funding to spend the resources on it.
Otherwise, you can just use Jitsi Video Bridge.

On Fri, Feb 19, 2016 at 10:31 AM, <martin.hoole at emailn.de> wrote:

> Are there plans to implement support for Selective Forwarding Unit (SFU)
> for video or audio rather than mixing video together?
> https://tools.ietf.org/html/draft-ietf-avtcore-rtp-topologies-update-01
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 deepikay at iiitd.ac.in  Fri Feb 19 20:09:30 2016
From: deepikay at iiitd.ac.in (Deepika Yadav)
Date: Fri, 19 Feb 2016 22:39:30 +0530
Subject: [Freeswitch-users] Library linking problem in ESL
Message-ID: <CAM99_Y=4NEsbUUMUy+wzrwcgUz7ZVJfCn=NsamG+Cz7yvR=8aw@mail.gmail.com>

Hi,

I am using java ESL, and getting the following error :

Exception in thread "main"
java.lang.UnsatisfiedLinkError:org.freeswitch.swig.freeswitchJNI.new_JavaSession_SWIG_0()J
        at org.freeswitch.swig.freeswitchJNI.new_JavaSession__SWIG_0(Native
Method)
        at org.freeswitch.swig.JavaSession.<init>(JavaSession.java:37)
        at MyESLTest.main(MyESLTest.java:465)

and I also activates another similar esl program from the dialplan that
runs fine.

would be thankful if any clue regarding resolving it can be suggested.

Regards,
Deepika
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From tomasz.o.ostrowski at gmail.com  Fri Feb 19 22:24:40 2016
From: tomasz.o.ostrowski at gmail.com (Tomasz Ostrowski)
Date: Fri, 19 Feb 2016 20:24:40 +0100
Subject: [Freeswitch-users] Incorrect reply to T.38 re-INVITE
Message-ID: <op.yc3hjevstbe8eu@ot.chello.pl>

Hello,
while testing FreeSWITCH 1.4.26 I stumbled upon old and seemingly ignored  
problem when FreeSWITCH replies with SDP containing:

m=image 0 udptl t38
m=image 0 udptl t38

when receiving re-INVITE with disabled audio media and enabled image media  
(switching from audio to image). As far as I know this is correct way to  
change media, from RFC 3264:

8.1 Adding a Media Stream

    New media streams are created by new additional media descriptions
    below the existing ones, or by reusing the "slot" used by an old
    media stream which had been disabled by setting its port to zero.

    Reusing its slot means that the new media description replaces the
    old one, but retains its positioning relative to other media
    descriptions in  the SDP.  New media descriptions MUST appear below
    any existing media sections.  The rules for formatting these media
    descriptions are identical to those described in Section 5.

    When the answerer receives an SDP with more media descriptions than
    the previous SDP from the offerer, or it receives an SDP with a media
    stream in a slot where the port was previously zero, the answerer
    knows that new media streams are being added.  These can be rejected
    or accepted by placing an appropriately structured media description
    in the answer.  The procedures for constructing the new media
    description in the answer are described in Section 6.

This problem is mentioned in:
https://freeswitch.org/jira/browse/FS-7037
https://freeswitch.org/jira/browse/FS-6212

With reversed transmission direction (when FreeSWITCH receives FAX and  
re-INVITES) re-INVITE contains only image media, but this is easier to  
accept (https://freeswitch.org/jira/browse/FS-6954 -  correcting it caused  
interoperability problems between FS versions), while not accepting  
correct SDP by FreeSWITCH is really painful as it requires implementing  
RFC non-compliant negotiation and probably adding special switch in  
configuration to be interoperable.

Is this issue fixed in FreeSWITCH 1.6 (I cannot find any further  
references in jira)?
Could you give me any suggestions where to look in FreeSWITCH source code  
(10k LOC sofia.c seems pretty complex)?

-- 
TMSZ


From Jerry.Kendall at FiberConX.com  Fri Feb 19 22:28:19 2016
From: Jerry.Kendall at FiberConX.com (Jerry Kendall)
Date: Fri, 19 Feb 2016 14:28:19 -0500
Subject: [Freeswitch-users] UNALLOCATED_NUMBER handling
Message-ID: <56C76CD3.3000008@FiberConX.com>

Hello all....

I'm trying to add code to the dialplan to handle the UNALLOCATED_NUMBER error.

I'm in Canada, when I call a US based 800 number I should, and now am getting UNALLOCATED_NUMBER 
error from the Canadian based ITSP.
However, I want to handle it in the DialPlan. I want to tell the users, via audio file playback, 
that the number is not assigned/allocated.

I have looked in lots of places and found references to transfer_on_fail and a few other ideas and 
have tried many but, I just can't seem to make this work.

Not sure if this is the best way to handle this.

my 'default' dialplan has this-------------
<include>
         <!--
                 North American Outbound call:
                 DID: 1NxxNxxXXXX
                 GW: GW-0
         -->
         <extension name="public_did_11234567890" continue="true">
                 <condition field="${caller_id_number}" expression="^11234567890$"/>
                 <condition field="${destination_number}" expression="^1?([2-9]\d{2}[2-9]\d{2}\d{4})$">

                     <action application="set" data="hangup_after_bridge=true"/>
                     <action application="set" data="continue_on_fail=true"/>
                     <action application="set" data="transfer_on_fail=UNALLOCATED_NUMBER 
UNALLOCATED_NUMBER xml error"/>

                     <action application="bridge" data="sofia/gateway/GW-0/1$1"/>

                 </condition>

         </extension>
</include>

and I have the following in for UNALLOCATED_NUMBER extension----
<include>
         <extension name="UNALLOCATED_NUMBER" continue="true">
                 <condition field="destination_number" expression="UNALLOCATED_NUMBER" 
continue="false" break="on-true">
                         <action application="sleep" data="1000"/>
                         <action application="playback" 
data="$${conf_dir}/sounds/unallocated_number.wav"/>
                         <action application="hangup" data="NORMAL_CLEARING"/>
                 </condition>
         </extension>
</include>



Jerry


From mike at jerris.com  Fri Feb 19 23:00:05 2016
From: mike at jerris.com (Michael Jerris)
Date: Fri, 19 Feb 2016 15:00:05 -0500
Subject: [Freeswitch-users] Incorrect reply to T.38 re-INVITE
In-Reply-To: <op.yc3hjevstbe8eu@ot.chello.pl>
References: <op.yc3hjevstbe8eu@ot.chello.pl>
Message-ID: <43B10980-E729-4087-ABA5-668C6F610A69@jerris.com>

I think we fixed this in 1.6 already.  Give it a try to confirm but i recall this one getting fixed.

> On Feb 19, 2016, at 2:24 PM, Tomasz Ostrowski <tomasz.o.ostrowski at gmail.com> wrote:
> 
> Hello,
> while testing FreeSWITCH 1.4.26 I stumbled upon old and seemingly ignored  
> problem when FreeSWITCH replies with SDP containing:
> 
> m=image 0 udptl t38
> m=image 0 udptl t38
> 
> when receiving re-INVITE with disabled audio media and enabled image media  
> (switching from audio to image). As far as I know this is correct way to  
> change media, from RFC 3264:
> 
> 8.1 Adding a Media Stream
> 
>    New media streams are created by new additional media descriptions
>    below the existing ones, or by reusing the "slot" used by an old
>    media stream which had been disabled by setting its port to zero.
> 
>    Reusing its slot means that the new media description replaces the
>    old one, but retains its positioning relative to other media
>    descriptions in  the SDP.  New media descriptions MUST appear below
>    any existing media sections.  The rules for formatting these media
>    descriptions are identical to those described in Section 5.
> 
>    When the answerer receives an SDP with more media descriptions than
>    the previous SDP from the offerer, or it receives an SDP with a media
>    stream in a slot where the port was previously zero, the answerer
>    knows that new media streams are being added.  These can be rejected
>    or accepted by placing an appropriately structured media description
>    in the answer.  The procedures for constructing the new media
>    description in the answer are described in Section 6.
> 
> This problem is mentioned in:
> https://freeswitch.org/jira/browse/FS-7037
> https://freeswitch.org/jira/browse/FS-6212
> 
> With reversed transmission direction (when FreeSWITCH receives FAX and  
> re-INVITES) re-INVITE contains only image media, but this is easier to  
> accept (https://freeswitch.org/jira/browse/FS-6954 -  correcting it caused  
> interoperability problems between FS versions), while not accepting  
> correct SDP by FreeSWITCH is really painful as it requires implementing  
> RFC non-compliant negotiation and probably adding special switch in  
> configuration to be interoperable.
> 
> Is this issue fixed in FreeSWITCH 1.6 (I cannot find any further  
> references in jira)?
> Could you give me any suggestions where to look in FreeSWITCH source code  
> (10k LOC sofia.c seems pretty complex)?
> 
> -- 
> TMSZ
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 nneul at mst.edu  Sat Feb 20 06:19:54 2016
From: nneul at mst.edu (Nathan Neulinger)
Date: Fri, 19 Feb 2016 21:19:54 -0600
Subject: [Freeswitch-users] one way audio - another report related to
 auto-jitter-buffer-msec possible behavior change with 1.6
Message-ID: <56C7DB5A.6030200@mst.edu>

Just wanted to add an additional data point on the one way audio issue (as it turns out - also with grandstreams) with 1.6.

	AnalogDevice <-> Grandstream <-> FS <-> SIP-Provider

One my most recent test/capture/etc, the behavior I saw was that it looked to me like audio stopped being sent from 
freeswitch to outside/external leg once early media finished. Note that the grandstream was continuing to send RTP to 
FS, but FS wasn't passing any of it on to the provider.

Once I saw the thread from the other user having this problem, tried turning it off on my profiles that had it enabled, 
and the problem appears to have gone away. I can provide logs/debug/captures/etc. if requested from a test environment.

I did not have any reports of this issue except related to grandstreams (ht-701) - in my case, almost all related to 
analog faxes (bulk of our ATA usage). Also have not been able to verify the fix yet on a call originating from the 
grandstream (testing from home).

-- Nathan

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


From anthony.minessale at gmail.com  Sat Feb 20 08:13:09 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Feb 2016 23:13:09 -0600
Subject: [Freeswitch-users] one way audio - another report related to
 auto-jitter-buffer-msec possible behavior change with 1.6
In-Reply-To: <56C7DB5A.6030200@mst.edu>
References: <56C7DB5A.6030200@mst.edu>
Message-ID: <CAKbxfG8jThcvOaZ_S3PY7fA4V6WTf+-sv1Zeya8Dt3Lc_yn3ew@mail.gmail.com>

Proported bugs and details therein belong on Jira.

On Friday, February 19, 2016, Nathan Neulinger <nneul at mst.edu> wrote:

> Just wanted to add an additional data point on the one way audio issue (as
> it turns out - also with grandstreams) with 1.6.
>
>         AnalogDevice <-> Grandstream <-> FS <-> SIP-Provider
>
> One my most recent test/capture/etc, the behavior I saw was that it looked
> to me like audio stopped being sent from
> freeswitch to outside/external leg once early media finished. Note that
> the grandstream was continuing to send RTP to
> FS, but FS wasn't passing any of it on to the provider.
>
> Once I saw the thread from the other user having this problem, tried
> turning it off on my profiles that had it enabled,
> and the problem appears to have gone away. I can provide
> logs/debug/captures/etc. if requested from a test environment.
>
> I did not have any reports of this issue except related to grandstreams
> (ht-701) - in my case, almost all related to
> analog faxes (bulk of our ATA usage). Also have not been able to verify
> the fix yet on a call originating from the
> grandstream (testing from home).
>
> -- Nathan
>
> ------------------------------------------------------------
> Nathan Neulinger                       nneul at mst.edu <javascript:;>
> Missouri S&T Information Technology    (573) 612-1412
> System Administrator - Architect
>
> _________________________________________________________________________
> 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 royj at yandex.ru  Sat Feb 20 09:26:42 2016
From: royj at yandex.ru (royj at yandex.ru)
Date: Sat, 20 Feb 2016 09:26:42 +0300
Subject: [Freeswitch-users] UNALLOCATED_NUMBER handling
In-Reply-To: <56C76CD3.3000008@FiberConX.com>
References: <56C76CD3.3000008@FiberConX.com>
Message-ID: <1442451455949602@web18o.yandex.ru>

I think it should work without that  'continue="true"' in 'extension name="public_did_11234567890"' and without 'hangup_after_bridge=true'

19.02.2016, 22:38, "Jerry Kendall" <jerry.kendall at fiberconx.com>:
> Hello all....
>
> I'm trying to add code to the dialplan to handle the UNALLOCATED_NUMBER error.
>
> I'm in Canada, when I call a US based 800 number I should, and now am getting UNALLOCATED_NUMBER
> error from the Canadian based ITSP.
> However, I want to handle it in the DialPlan. I want to tell the users, via audio file playback,
> that the number is not assigned/allocated.
>
> I have looked in lots of places and found references to transfer_on_fail and a few other ideas and
> have tried many but, I just can't seem to make this work.
>
> Not sure if this is the best way to handle this.
>
> my 'default' dialplan has this-------------
> <include>
> ?????????<!--
> ?????????????????North American Outbound call:
> ?????????????????DID: 1NxxNxxXXXX
> ?????????????????GW: GW-0
> ?????????-->
> ?????????<extension name="public_did_11234567890" continue="true">
> ?????????????????<condition field="${caller_id_number}" expression="^11234567890$"/>
> ?????????????????<condition field="${destination_number}" expression="^1?([2-9]\d{2}[2-9]\d{2}\d{4})$">
>
> ?????????????????????<action application="set" data="hangup_after_bridge=true"/>
> ?????????????????????<action application="set" data="continue_on_fail=true"/>
> ?????????????????????<action application="set" data="transfer_on_fail=UNALLOCATED_NUMBER
> UNALLOCATED_NUMBER xml error"/>
>
> ?????????????????????<action application="bridge" data="sofia/gateway/GW-0/1$1"/>
>
> ?????????????????</condition>
>
> ?????????</extension>
> </include>
>
> and I have the following in for UNALLOCATED_NUMBER extension----
> <include>
> ?????????<extension name="UNALLOCATED_NUMBER" continue="true">
> ?????????????????<condition field="destination_number" expression="UNALLOCATED_NUMBER"
> continue="false" break="on-true">
> ?????????????????????????<action application="sleep" data="1000"/>
> ?????????????????????????<action application="playback"
> data="$${conf_dir}/sounds/unallocated_number.wav"/>
> ?????????????????????????<action application="hangup" data="NORMAL_CLEARING"/>
> ?????????????????</condition>
> ?????????</extension>
> </include>
>
> Jerry
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 s.safarov at gmail.com  Sat Feb 20 12:44:08 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 20 Feb 2016 12:44:08 +0300
Subject: [Freeswitch-users] new version of vpx needed for those tracking
	master.
In-Reply-To: <6F83920F-35F5-4B15-B9CC-AC5C4648C7DA@jerris.com>
References: <6F83920F-35F5-4B15-B9CC-AC5C4648C7DA@jerris.com>
Message-ID: <CAHtxdDdYAG3gRxSZZJKx0du+PwZVGRkbz-Zg8ty-sYFOr5GPgA@mail.gmail.com>

Creaed PR4
<https://freeswitch.org/stash/projects/SD/repos/libvpx/pull-requests/4/overview>
to allow rpm package creation

On Thu, Feb 18, 2016 at 7:28 PM, Michael Jerris <mike at jerris.com> wrote:

> Please note, we have updated code in master that will now require the
> newer version of libvpx (and anything such as libav that is linked against
> vpx will also need to be updated) in order to use video features.  The
> tarball for this newer version is located at
> http://files.freeswitch.org/downloads/libs/libvpx2-1.5.0.tar.gz .  If you
> are using our debian packages for dependencies, these packages are already
> available in the unstable repo.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 bilaln018 at gmail.com  Sat Feb 20 16:08:29 2016
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Sat, 20 Feb 2016 18:08:29 +0500
Subject: [Freeswitch-users] [mod_skyopen][Debian Jessie][Error while loading]
Message-ID: <CADSVerCof2q4e5eTgViHLDjGtZD3re1yorBrgt=jzdDkCXc5sQ@mail.gmail.com>

Hi Users,
I am getting error while loading mod_skyopen on debian jessie 8 64bit, i am
using freeswitch Version 1.7.0 git 09b4156 2016-02-04 17:08:30Z 64bit

Logs: https://pastebin.freeswitch.org/24569

I could not find any official link for installation on Debian 8.

Regards
Abbasi
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From bilaln018 at gmail.com  Sat Feb 20 16:09:31 2016
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Sat, 20 Feb 2016 18:09:31 +0500
Subject: [Freeswitch-users] Skypopen Debian Jessie install issues
In-Reply-To: <CAHtxdDdg8_M3FFd2MaF1STvZv5CZOAgkdBy0JO1xQ=NuLy77eg@mail.gmail.com>
References: <CANHADwO9=X+CzVmH6p6GrngGGp+2ssez1sk0aStAj+_-tKQFDA@mail.gmail.com>
	<CAHtxdDdg8_M3FFd2MaF1STvZv5CZOAgkdBy0JO1xQ=NuLy77eg@mail.gmail.com>
Message-ID: <CADSVerC3W4Aa9y9T56Hj=UikMcxFPjoCORM9R3v4M9vTE+ei_A@mail.gmail.com>

Did you got any fix?
I am facing the same.

Regards
Abbasi

On Wed, Dec 16, 2015 at 12:50 AM, Sergey Safarov <s.safarov at gmail.com>
wrote:

> Please read
> http://lists.freeswitch.org/pipermail/freeswitch-users/2015-November/117273.html
>
> On Tue, Dec 15, 2015 at 3:08 PM, Jason Bedward <mase2hot at gmail.com> wrote:
>
>> Hi,
>>
>> I have tried to install this on Debian Jessie but had some issues with
>> dependencies which are not matching what the install guide says. When I got
>> to the CLI and load the module I get the error below. I
>> have libjpeg62-turbo & libjpeg62-turbo-dev installed which I think it the
>> current version for Debian. Please can someone help guide me with this
>> version of Debian.
>>
>> Thanks
>>
>>
>> +OK log level 9 [9]
>> +OK console log level set to DEBUG
>>
>> freeswitch at 192.168.2.233@internal> load mod_skypopen
>> +OK Reloading XML
>> -ERR [module load file routine returned an error]
>>
>> 2015-12-15 12:01:55.996236 [INFO] mod_enum.c:880 ENUM Reloaded
>> 2015-12-15 12:01:55.996236 [INFO] switch_time.c:1415 Timezone reloaded
>> 1781 definitions
>> 2015-12-15 12:01:55.996236 [CRIT] switch_loadable_module.c:1447 Error
>> Loading module /usr/local/freeswitch/mod/mod_skypopen.so
>> **/usr/local/freeswitch/mod/mod_skypopen.so: undefined symbol:
>> jpeg_resync_to_restart**
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From nneul at mst.edu  Sat Feb 20 16:48:31 2016
From: nneul at mst.edu (Nathan Neulinger)
Date: Sat, 20 Feb 2016 07:48:31 -0600
Subject: [Freeswitch-users] one way audio - another report related to
 auto-jitter-buffer-msec possible behavior change with 1.6
In-Reply-To: <CAKbxfG8jThcvOaZ_S3PY7fA4V6WTf+-sv1Zeya8Dt3Lc_yn3ew@mail.gmail.com>
References: <56C7DB5A.6030200@mst.edu>
	<CAKbxfG8jThcvOaZ_S3PY7fA4V6WTf+-sv1Zeya8Dt3Lc_yn3ew@mail.gmail.com>
Message-ID: <56C86EAF.9060402@mst.edu>

https://freeswitch.org/jira/browse/FS-8850

On 02/19/2016 11:13 PM, Anthony Minessale wrote:
> Proported bugs and details therein belong on Jira.
>
> On Friday, February 19, 2016, Nathan Neulinger <nneul at mst.edu <mailto:nneul at mst.edu>> wrote:
>
>     Just wanted to add an additional data point on the one way audio issue (as it turns out - also with grandstreams)
>     with 1.6.
>
>              AnalogDevice <-> Grandstream <-> FS <-> SIP-Provider
>
>     One my most recent test/capture/etc, the behavior I saw was that it looked to me like audio stopped being sent from
>     freeswitch to outside/external leg once early media finished. Note that the grandstream was continuing to send RTP to
>     FS, but FS wasn't passing any of it on to the provider.
>
>     Once I saw the thread from the other user having this problem, tried turning it off on my profiles that had it enabled,
>     and the problem appears to have gone away. I can provide logs/debug/captures/etc. if requested from a test environment.
>
>     I did not have any reports of this issue except related to grandstreams (ht-701) - in my case, almost all related to
>     analog faxes (bulk of our ATA usage). Also have not been able to verify the fix yet on a call originating from the
>     grandstream (testing from home).
>
>     -- Nathan
>
>     ------------------------------------------------------------
>     Nathan Neulinger nneul at mst.edu <javascript:;>
>     Missouri S&T Information Technology    (573) 612-1412
>     System Administrator - Architect
>
>     _________________________________________________________________________
>     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 <http://irc.freenode.net> #freeswitch ? _http://freeswitch.org/g+_
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org <mailto:sip%3A888 at conference.freeswitch.org>  ? +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 s.safarov at gmail.com  Sat Feb 20 17:24:01 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 20 Feb 2016 17:24:01 +0300
Subject: [Freeswitch-users] recording problems with mod_shout
	debian/master
In-Reply-To: <CAEvFzR8Xxu57=rLnbwPkMJFAGMJXn7O2cik41YuFv128BcwZ9g@mail.gmail.com>
References: <CAEvFzR8Xxu57=rLnbwPkMJFAGMJXn7O2cik41YuFv128BcwZ9g@mail.gmail.com>
Message-ID: <CAHtxdDccgrBTnJJr+e4P+e1onwc3TgBJrt9Hh6XmbrEWER72Bw@mail.gmail.com>

Detailed description of error is given on FS-8851
<https://freeswitch.org/jira/browse/FS-8851>
Sergey

On Mon, Feb 1, 2016 at 4:47 PM, Luis Azedo <luis.azedo at factorlusitano.com>
wrote:

> Hi,
>
> anyone having problems recording in mp3  format ?
>
> 2016-02-01 13:41:47.108078 [INFO] kazoo_node.c:625 exec: *uuid_record(96722133-5060-508 at BJC.BGI.CG.BD
> <96722133-5060-508 at BJC.BGI.CG.BD> start
> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3 600)*
> 2016-02-01 13:41:47.108078 [DEBUG]* switch_core_file.c:323 File
> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3 sample rate 8000 doesn't match
> requested rate 16000*
> 2016-02-01 13:41:47.128429 [DEBUG] *switch_core_media_bug.c:828 Attaching
> BUG to sofia/sipinterface_1/995582142 at teste.sip.90e9.com
> <995582142 at teste.sip.90e9.com>*
> 2016-02-01 13:41:47.148020 [DEBUG] *switch_ivr_async.c:1491 No silence
> detection configured; assuming start of speech*
> 2016-02-01 13:41:47.168888 [INFO] *mod_shout.c:326 LAME 3.99.5 64bits
> (http://lame.sf.net <http://lame.sf.net>)*
> 2016-02-01 13:41:47.168888 [INFO] *mod_shout.c:326 polyphase lowpass
> filter disabled*
> 2016-02-01 13:41:47.168888 [ERR] *mod_shout.c:1057 MP3 encode error -1!*
> 2016-02-01 13:41:47.168888 [ERR] *switch_ivr_async.c:1160 Error writing
> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3*
> 2016-02-01 13:41:47.168888 [ERR] *mod_shout.c:1057 MP3 encode error -1!*
> 2016-02-01 13:41:47.168888 [ERR] *switch_ivr_async.c:1160 Error writing
> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3*
> 2016-02-01 13:41:47.688430 [ERR] *mod_shout.c:1057 MP3 encode error -1!*
> 2016-02-01 13:41:47.688430 [ERR] *switch_ivr_async.c:1160 Error writing
> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3*
> 2016-02-01 13:41:47.688430 [ERR] *mod_shout.c:1057 MP3 encode error -1!*
> 2016-02-01 13:41:47.688430 [ERR] *switch_ivr_async.c:1160 Error writing
> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3*
> 2016-02-01 13:41:47.688430 [ERR] *mod_shout.c:1057 MP3 encode error -1!*
> 2016-02-01 13:41:47.688430 [ERR] *switch_ivr_async.c:1160 Error writing
> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3*
>
> 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 anthony.minessale at gmail.com  Sat Feb 20 20:08:49 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 20 Feb 2016 11:08:49 -0600
Subject: [Freeswitch-users] recording problems with mod_shout
	debian/master
In-Reply-To: <CAHtxdDccgrBTnJJr+e4P+e1onwc3TgBJrt9Hh6XmbrEWER72Bw@mail.gmail.com>
References: <CAEvFzR8Xxu57=rLnbwPkMJFAGMJXn7O2cik41YuFv128BcwZ9g@mail.gmail.com>
	<CAHtxdDccgrBTnJJr+e4P+e1onwc3TgBJrt9Hh6XmbrEWER72Bw@mail.gmail.com>
Message-ID: <CAKbxfG-eDZm+qRuSPg4mJXedrc1kVmMg+ZfJrf39t_KVi7VO-w@mail.gmail.com>

Does kazoo have bug reports?
They are probably starting the call recording too soon.

On Saturday, February 20, 2016, Sergey Safarov <s.safarov at gmail.com> wrote:

> Detailed description of error is given on FS-8851
> <https://freeswitch.org/jira/browse/FS-8851>
> Sergey
>
> On Mon, Feb 1, 2016 at 4:47 PM, Luis Azedo <luis.azedo at factorlusitano.com
> <javascript:_e(%7B%7D,'cvml','luis.azedo at factorlusitano.com');>> wrote:
>
>> Hi,
>>
>> anyone having problems recording in mp3  format ?
>>
>> 2016-02-01 13:41:47.108078 [INFO] kazoo_node.c:625 exec: *uuid_record(96722133-5060-508 at BJC.BGI.CG.BD
>> <javascript:_e(%7B%7D,'cvml','96722133-5060-508 at BJC.BGI.CG.BD');> start
>> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3 600)*
>> 2016-02-01 13:41:47.108078 [DEBUG]* switch_core_file.c:323 File
>> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3 sample rate 8000 doesn't match
>> requested rate 16000*
>> 2016-02-01 13:41:47.128429 [DEBUG] *switch_core_media_bug.c:828
>> Attaching BUG to sofia/sipinterface_1/995582142 at teste.sip.90e9.com
>> <javascript:_e(%7B%7D,'cvml','995582142 at teste.sip.90e9.com');>*
>> 2016-02-01 13:41:47.148020 [DEBUG] *switch_ivr_async.c:1491 No silence
>> detection configured; assuming start of speech*
>> 2016-02-01 13:41:47.168888 [INFO] *mod_shout.c:326 LAME 3.99.5 64bits
>> (http://lame.sf.net <http://lame.sf.net>)*
>> 2016-02-01 13:41:47.168888 [INFO] *mod_shout.c:326 polyphase lowpass
>> filter disabled*
>> 2016-02-01 13:41:47.168888 [ERR] *mod_shout.c:1057 MP3 encode error -1!*
>> 2016-02-01 13:41:47.168888 [ERR] *switch_ivr_async.c:1160 Error writing
>> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3*
>> 2016-02-01 13:41:47.168888 [ERR] *mod_shout.c:1057 MP3 encode error -1!*
>> 2016-02-01 13:41:47.168888 [ERR] *switch_ivr_async.c:1160 Error writing
>> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3*
>> 2016-02-01 13:41:47.688430 [ERR] *mod_shout.c:1057 MP3 encode error -1!*
>> 2016-02-01 13:41:47.688430 [ERR] *switch_ivr_async.c:1160 Error writing
>> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3*
>> 2016-02-01 13:41:47.688430 [ERR] *mod_shout.c:1057 MP3 encode error -1!*
>> 2016-02-01 13:41:47.688430 [ERR] *switch_ivr_async.c:1160 Error writing
>> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3*
>> 2016-02-01 13:41:47.688430 [ERR] *mod_shout.c:1057 MP3 encode error -1!*
>> 2016-02-01 13:41:47.688430 [ERR] *switch_ivr_async.c:1160 Error writing
>> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3*
>>
>> Thanks
>>
>>
>> _________________________________________________________________________
>> 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  ? +19193869900

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From d at d-man.org  Sat Feb 20 20:23:10 2016
From: d at d-man.org (Darren)
Date: Sat, 20 Feb 2016 17:23:10 +0000
Subject: [Freeswitch-users] recording problems with mod_shout
 debian/master
In-Reply-To: <CAKbxfG-eDZm+qRuSPg4mJXedrc1kVmMg+ZfJrf39t_KVi7VO-w@mail.gmail.com>
References: <CAEvFzR8Xxu57=rLnbwPkMJFAGMJXn7O2cik41YuFv128BcwZ9g@mail.gmail.com>
	<CAHtxdDccgrBTnJJr+e4P+e1onwc3TgBJrt9Hh6XmbrEWER72Bw@mail.gmail.com>
	<CAKbxfG-eDZm+qRuSPg4mJXedrc1kVmMg+ZfJrf39t_KVi7VO-w@mail.gmail.com>
Message-ID: <77E4928D-EC8E-40D9-97FD-9562B6AA0B26@d-man.org>

Can you please file this at tickets.2600hz.com and we'll take a look? This seems odd to me.

From: <freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>> on behalf of Anthony Minessale <anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Date: Sunday, February 21, 2016 at 6:08 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] recording problems with mod_shout debian/master

Does kazoo have bug reports?
They are probably starting the call recording too soon.

On Saturday, February 20, 2016, Sergey Safarov <s.safarov at gmail.com<mailto:s.safarov at gmail.com>> wrote:
Detailed description of error is given on FS-8851<https://freeswitch.org/jira/browse/FS-8851>
Sergey

On Mon, Feb 1, 2016 at 4:47 PM, Luis Azedo <luis.azedo at factorlusitano.com<javascript:_e(%7B%7D,'cvml','luis.azedo at factorlusitano.com');>> wrote:
Hi,

anyone having problems recording in mp3  format ?

2016-02-01 13:41:47.108078 [INFO] kazoo_node.c:625 exec: uuid_record(96722133-5060-508 at BJC.BGI.CG.BD<javascript:_e(%7B%7D,'cvml','96722133-5060-508 at BJC.BGI.CG.BD');> start /tmp/878202b3a78863ec40f2dbc800fcde66.mp3 600)
2016-02-01 13:41:47.108078 [DEBUG] switch_core_file.c:323 File /tmp/878202b3a78863ec40f2dbc800fcde66.mp3 sample rate 8000 doesn't match requested rate 16000
2016-02-01 13:41:47.128429 [DEBUG] switch_core_media_bug.c:828 Attaching BUG to sofia/sipinterface_1/995582142 at teste.sip.90e9.com<javascript:_e(%7B%7D,'cvml','995582142 at teste.sip.90e9.com');>
2016-02-01 13:41:47.148020 [DEBUG] switch_ivr_async.c:1491 No silence detection configured; assuming start of speech
2016-02-01 13:41:47.168888 [INFO] mod_shout.c:326 LAME 3.99.5 64bits (http://lame.sf.net)
2016-02-01 13:41:47.168888 [INFO] mod_shout.c:326 polyphase lowpass filter disabled
2016-02-01 13:41:47.168888 [ERR] mod_shout.c:1057 MP3 encode error -1!
2016-02-01 13:41:47.168888 [ERR] switch_ivr_async.c:1160 Error writing /tmp/878202b3a78863ec40f2dbc800fcde66.mp3
2016-02-01 13:41:47.168888 [ERR] mod_shout.c:1057 MP3 encode error -1!
2016-02-01 13:41:47.168888 [ERR] switch_ivr_async.c:1160 Error writing /tmp/878202b3a78863ec40f2dbc800fcde66.mp3
2016-02-01 13:41:47.688430 [ERR] mod_shout.c:1057 MP3 encode error -1!
2016-02-01 13:41:47.688430 [ERR] switch_ivr_async.c:1160 Error writing /tmp/878202b3a78863ec40f2dbc800fcde66.mp3
2016-02-01 13:41:47.688430 [ERR] mod_shout.c:1057 MP3 encode error -1!
2016-02-01 13:41:47.688430 [ERR] switch_ivr_async.c:1160 Error writing /tmp/878202b3a78863ec40f2dbc800fcde66.mp3
2016-02-01 13:41:47.688430 [ERR] mod_shout.c:1057 MP3 encode error -1!
2016-02-01 13:41:47.688430 [ERR] switch_ivr_async.c:1160 Error writing /tmp/878202b3a78863ec40f2dbc800fcde66.mp3

Thanks


_________________________________________________________________________
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<http://irc.freenode.net> #freeswitch ? http://freeswitch.org/g+

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA

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From gregor at infomedia.si  Sun Feb 21 02:00:25 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Sun, 21 Feb 2016 00:00:25 +0100
Subject: [Freeswitch-users] Verto
Message-ID: <CAC7ttFF_fL+md9Zf25OfM3_BB6Dwyto09YVxNq88Cgid-RG8aQ@mail.gmail.com>

Hi!

Haven't try Verto yet, but just wondering what happens if I login in
multiple browser window with same username? What will happen on outgoing or
incoming call?

Best regards, Gregor
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From krice at freeswitch.org  Sun Feb 21 04:58:40 2016
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 20 Feb 2016 19:58:40 -0600
Subject: [Freeswitch-users] Verto
In-Reply-To: <CAC7ttFF_fL+md9Zf25OfM3_BB6Dwyto09YVxNq88Cgid-RG8aQ@mail.gmail.com>
References: <CAC7ttFF_fL+md9Zf25OfM3_BB6Dwyto09YVxNq88Cgid-RG8aQ@mail.gmail.com>
Message-ID: <23ca01d16c4b$63206730$29613590$@freeswitch.org>

This depends on several factors? but it is workable in many configs this way?.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gregor Nanger
Sent: Saturday, February 20, 2016 5:00 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Verto

 

Hi!

 

Haven't try Verto yet, but just wondering what happens if I login in multiple browser window with same username? What will happen on outgoing or incoming call?

 

Best regards, Gregor

 

 

  <http://t.sidekickopen35.com/e1t/o/5/f18dQhb0S7ks8dDMPbW2n0x6l2B9gXrN7sKj6v5dbp0W3MPnzY4XrbSMW8q-f_43LvrVvW59BB4r1k1H6H0?si=5036397111279616&pi=d36c57d6-b3c6-4ec8-9c03-0371561ce6f9> 

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From s.safarov at gmail.com  Sun Feb 21 06:26:57 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sun, 21 Feb 2016 03:26:57 +0000
Subject: [Freeswitch-users] Free learning WebRTC book
Message-ID: <CAHtxdDdoXXSZ0jkQ33f0E+UDiwC4iLnCZsntEy1DYzNX1N68NQ@mail.gmail.com>

If you interested, please download WebRTC book
https://www.packtpub.com/packt/offers/free-learning
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From gregor at infomedia.si  Sun Feb 21 09:17:46 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Sun, 21 Feb 2016 07:17:46 +0100
Subject: [Freeswitch-users] Verto
In-Reply-To: <23ca01d16c4b$63206730$29613590$@freeswitch.org>
References: <CAC7ttFF_fL+md9Zf25OfM3_BB6Dwyto09YVxNq88Cgid-RG8aQ@mail.gmail.com>
	<23ca01d16c4b$63206730$29613590$@freeswitch.org>
Message-ID: <CAC7ttFFbvOApx2Ci_n9iUmzzNLOHeAysBeipJt3H0dEDnTOdow@mail.gmail.com>

Guess it is same as login with softphone with same credentials. It would be
best to try and see  :-)

2016-02-21 2:58 GMT+01:00 Ken Rice <krice at freeswitch.org>:

> This depends on several factors? but it is workable in many configs this
> way?.
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Gregor
> Nanger
> *Sent:* Saturday, February 20, 2016 5:00 PM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] Verto
>
>
>
> Hi!
>
>
>
> Haven't try Verto yet, but just wondering what happens if I login in
> multiple browser window with same username? What will happen on outgoing or
> incoming call?
>
>
>
> 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
>



-- 
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  Sun Feb 21 09:31:25 2016
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 21 Feb 2016 00:31:25 -0600
Subject: [Freeswitch-users] Verto
In-Reply-To: <CAC7ttFFbvOApx2Ci_n9iUmzzNLOHeAysBeipJt3H0dEDnTOdow@mail.gmail.com>
References: <CAC7ttFF_fL+md9Zf25OfM3_BB6Dwyto09YVxNq88Cgid-RG8aQ@mail.gmail.com>
	<23ca01d16c4b$63206730$29613590$@freeswitch.org>
	<CAC7ttFFbvOApx2Ci_n9iUmzzNLOHeAysBeipJt3H0dEDnTOdow@mail.gmail.com>
Message-ID: <798CC583-BA4F-4CE1-BA83-264146ACC965@freeswitch.org>

My cryptic reply wasnt meant to be cryptic, its just it really depends on his configs and how he sets it up

Same browser different tabs works differently if depending on old school standard move vs private browsing mode and may depend on how the things are even configured... His question doesnt have enough specifics for a direct answer 

Sent from my iPhone

> On Feb 21, 2016, at 12:17 AM, Gregor Nanger <gregor at infomedia.si> wrote:
> 
> Guess it is same as login with softphone with same credentials. It would be best to try and see  :-)
> 
> 2016-02-21 2:58 GMT+01:00 Ken Rice <krice at freeswitch.org>:
>> This depends on several factors? but it is workable in many configs this way?.
>> 
>>  
>> 
>>  
>> 
>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gregor Nanger
>> Sent: Saturday, February 20, 2016 5:00 PM
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Subject: [Freeswitch-users] Verto
>> 
>>  
>> 
>> Hi!
>> 
>>  
>> 
>> Haven't try Verto yet, but just wondering what happens if I login in multiple browser window with same username? What will happen on outgoing or incoming call?
>> 
>>  
>> 
>> 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
> 
> 
> 
> -- 
> 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 alhakeem at gmail.com  Sun Feb 21 15:34:09 2016
From: alhakeem at gmail.com (Abdul Hakeem)
Date: Sun, 21 Feb 2016 12:34:09 +0000
Subject: [Freeswitch-users] Verto & Sofia Registrations
Message-ID: <CAO=waFDPZOypQdPba7NfF4vsn2TPoRFMCMqYT+jCMJGrLeuSNQ@mail.gmail.com>

Hello,
My understanding is that Sofia can only do one registration at a time,
does Veryo exhibit this limit ?

Is there any way to process multiple registrations in the background or to
configure FS to use the database of another REGISTRAR ?.

Much obliged,

AH
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From adahary at gmail.com  Sun Feb 21 15:58:42 2016
From: adahary at gmail.com (Assaf Dahary)
Date: Sun, 21 Feb 2016 14:58:42 +0200
Subject: [Freeswitch-users] Freeswitch extended with ZRTP
Message-ID: <030f01d16ca7$98240250$c86c06f0$@gmail.com>

Hi,

 

I would like to setup FS1 as a 'ZRTP gateway' to a none-zrtp sip client.

 

Here is my net setup: 

Client1 (without ZRTP)-> LAN -> FS1 (Gateway: register to FS2) -> NAT ->
Internet-> Public IP (not NAT) FS2 -> CSipSimple (ZRTP enabled).

 

Client1:

?  Ex#1000, Registered over LAN to FS1.

FS1:

?  Being NAT with dynamic IP address

?  Registered as a Gateway to FS2 (in Internal profile). So FS1 is extended
on FS2.

?  Setup as 'Proxy-Media = false' && zrtp_enrollment=true (trusted MITM).

FS2:

?  Connected with static public IP address (not behind NAT)

?  Setup as 'Proxy-Media = true' && inbound-late-negotiation=true. 

CSipSimple:

?  Behind NAT (remote WiFi/3G) and is registered on FS2 (Internal profile).

FS2 is successfully serving multiple CSipSimple ZRTP clients with end-to-end
ZRTP secure calls.

 

The problem:

When calling from Client1/FS1 to FS2/CSipSimple, 

then FS1 shows ' WARNING! Incoming ZRTP CRC validation fails' 

and FS2 shows ' ZRTP not negotiated on both sides; disabling ZRTP passthr
'.

 

I have tested FS1 locally with CSipSimple and it manages to connect with
ZRTP/MITM so it is capable of ZRTP.

 

I follow up what have been recommended on the forum to extend FS1 and to
verify matching codecs (I forced PCMU only on all devices) and to check
RTP/UDP ports flow (SIP trace).

 

I would appreciate any help on how to setup end-to-end ZRTP calls between
FS1 and remote CsipSimple (FS2).

 

Regards

 

Assaf  

 

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From krice at freeswitch.org  Sun Feb 21 19:07:51 2016
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 21 Feb 2016 10:07:51 -0600
Subject: [Freeswitch-users] Verto & Sofia Registrations
In-Reply-To: <CAO=waFDPZOypQdPba7NfF4vsn2TPoRFMCMqYT+jCMJGrLeuSNQ@mail.gmail.com>
References: <CAO=waFDPZOypQdPba7NfF4vsn2TPoRFMCMqYT+jCMJGrLeuSNQ@mail.gmail.com>
Message-ID: <D05681A9-5DBA-4B12-8B6B-76CDFE761C0D@freeswitch.org>

Sofia can do multi-reg its a setting on the profile 

Sent from my iPhone

> On Feb 21, 2016, at 6:34 AM, Abdul Hakeem <alhakeem at gmail.com> wrote:
> 
> Hello,
> My understanding is that Sofia can only do one registration at a time,  does Veryo exhibit this limit ?
> 
> Is there any way to process multiple registrations in the background or to configure FS to use the database of another REGISTRAR ?.
> 
> Much obliged,
> 
> 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 gregor at infomedia.si  Sun Feb 21 23:19:20 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Sun, 21 Feb 2016 21:19:20 +0100
Subject: [Freeswitch-users] Verto
In-Reply-To: <798CC583-BA4F-4CE1-BA83-264146ACC965@freeswitch.org>
References: <CAC7ttFF_fL+md9Zf25OfM3_BB6Dwyto09YVxNq88Cgid-RG8aQ@mail.gmail.com>
	<23ca01d16c4b$63206730$29613590$@freeswitch.org>
	<CAC7ttFFbvOApx2Ci_n9iUmzzNLOHeAysBeipJt3H0dEDnTOdow@mail.gmail.com>
	<798CC583-BA4F-4CE1-BA83-264146ACC965@freeswitch.org>
Message-ID: <CAC7ttFHrxwDprcOt1P8QCoEvXANoiFd6VrSnSkxK9jMNAp=mWQ@mail.gmail.com>

Thank you Ken, it is ok... I just have to try to have more specific
question.

2016-02-21 7:31 GMT+01:00 Ken Rice <krice at freeswitch.org>:

> My cryptic reply wasnt meant to be cryptic, its just it really depends on
> his configs and how he sets it up
>
> Same browser different tabs works differently if depending on old school
> standard move vs private browsing mode and may depend on how the things are
> even configured... His question doesnt have enough specifics for a direct
> answer
>
> Sent from my iPhone
>
> On Feb 21, 2016, at 12:17 AM, Gregor Nanger <gregor at infomedia.si> wrote:
>
> Guess it is same as login with softphone with same credentials. It would
> be best to try and see  :-)
>
> 2016-02-21 2:58 GMT+01:00 Ken Rice <krice at freeswitch.org>:
>
>> This depends on several factors? but it is workable in many configs this
>> way?.
>>
>>
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Gregor
>> Nanger
>> *Sent:* Saturday, February 20, 2016 5:00 PM
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* [Freeswitch-users] Verto
>>
>>
>>
>> Hi!
>>
>>
>>
>> Haven't try Verto yet, but just wondering what happens if I login in
>> multiple browser window with same username? What will happen on outgoing or
>> incoming call?
>>
>>
>>
>> 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
>>
>
>
>
> --
> 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 Harald.Petrovitsch at sermotec.at  Sun Feb 21 23:57:00 2016
From: Harald.Petrovitsch at sermotec.at (Harald Petrovitsch)
Date: Sun, 21 Feb 2016 20:57:00 +0000
Subject: [Freeswitch-users] Freeswitch with freetdm on Windows
Message-ID: <ab022529c23845808b00b0c9e9ea201c@imladris.sermotec.local>

Hi,

is anybody out there using Freeswitch with Sangoma ISDN E1 Hardware on Windows in a productive environment ?
Any stability Issues ? 
Which Freeswitch and Wanpipe version are you using ?

Regards
Harald

-- 
harald.petrovitsch at sermotec.at


From nandy1925 at gmail.com  Mon Feb 22 01:24:05 2016
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Mon, 22 Feb 2016 06:24:05 +0800
Subject: [Freeswitch-users] Free learning WebRTC book
In-Reply-To: <CAHtxdDdoXXSZ0jkQ33f0E+UDiwC4iLnCZsntEy1DYzNX1N68NQ@mail.gmail.com>
References: <CAHtxdDdoXXSZ0jkQ33f0E+UDiwC4iLnCZsntEy1DYzNX1N68NQ@mail.gmail.com>
Message-ID: <CAHKqMb7spOA+cC32W-7SpgoKy1maj-FJ=7ZDC8Hz7_T2-cjfpg@mail.gmail.com>

Thanks Sergey :-)

On Sun, Feb 21, 2016 at 11:26 AM, Sergey Safarov <s.safarov at gmail.com>
wrote:

> If you interested, please download WebRTC book
> https://www.packtpub.com/packt/offers/free-learning
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 rutu.patel at inextrix.com  Mon Feb 22 09:19:29 2016
From: rutu.patel at inextrix.com (Rutu Patel)
Date: Mon, 22 Feb 2016 11:49:29 +0530
Subject: [Freeswitch-users] Call dropping after 32 seconds
Message-ID: <CAPQet1zs08bGNnGDVmsLH3R3VbvMCJW=8Cx3U+KPxwGpYSuzAA@mail.gmail.com>

 Hello,

Having issue of call dropping after 32 seconds, here are the details-

x.x.x.174: opensips server
x.x.x.166: freeswitch server
x.x.x.3:     another opensips server which is registered as gateway on
above freeswitch server
x.x.x.6: freeswitch server
x.x.x.47:  server through which the user is registered
I am trying to call from xxxx9 to xxxxxxx29858
xxxxxxx00181 is caller-id name and caller-id number

Call flow is like this:
registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips
server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server)

Outbound-proxy is set to x.x.x.174 in Gateway configuration.

1) Call is initiated by the user(xxxx9) registered with host x.x.x.174
2) Call hit the freeswitch server x.x.x.166
3) After '180 Ringing' and '183 Session Progress' packet sending-receiving
started between 'x.x.x.174' and 'x.x.x.166' through the gateway x.x.x.3
But after 32 seconds call is dropped,
Within 32 seconds audio is ok from both end so it should not be the RTP
issue.
Here I have attached the file with sip logs, you can observer from the file
that, there are many '200 OK' from x.x.x.174 to x.x.x.166
and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then 'BYE'
from x.x.x.166 to x.x.x.174 and call is dropped.

What is wrong here? Any help would be appreciated here.

Here is the file with sip logs
--
Thanks,
Rutu Patel
<http://www.inextrix.com>
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U x.x.x.174:5060 -> x.x.x.166:5060
INVITE sip:xxxxxxx29858 at x.x.x.166:5060 SIP/2.0.
Record-Route: <sip:x.x.x.174:5060;lr;ftag=1037646889;did=3dd.18dd4fd7>.
Via: SIP/2.0/UDP x.x.x.174:5060;branch=z9hG4bK1bf1.f4e1b863.0.
Via: SIP/2.0/UDP x.x.x.47:2313;received=x.x.x.47;branch=z9hG4bK1606798992;rport=2313.
From: <sip:xxxx9 at x.x.x.174>;tag=1037646889.
To: <sip:xxxxxxx29858 at x.x.x.174>.
Call-ID: 1770373286-5062-8 at BJC.BGI.B.BFJ.
CSeq: 70 INVITE.
Contact: <sip:xxxx9 at x.x.x.47:2313>.
Max-Forwards: 30.
User-Agent: Grandstream GXP1400 1.0.6.11.
Privacy: none.
P-Preferred-Identity: <sip:xxxx9 at x.x.x.174>.
Supported: replaces, path, timer.
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE.
Content-Type: application/sdp.
Accept: application/sdp, application/dtmf-relay.
Content-Length:   285.
X-AUTH-IP: x.x.x.47.
.
v=0.
o=xxxx9 8001 8000 IN IP4 x.x.x.47.
s=SIP Call.
c=IN IP4 x.x.x.47.
t=0 0.
m=audio 5008 RTP/AVP 0 8 18 101.
a=sendrecv.
a=rtpmap:0 PCMU/8000.
a=ptime:20.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.


U x.x.x.166:5060 -> x.x.x.174:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP x.x.x.174:5060;branch=z9hG4bK1bf1.f4e1b863.0.
Via: SIP/2.0/UDP x.x.x.47:2313;received=x.x.x.47;branch=z9hG4bK1606798992;rport=2313.
Record-Route: <sip:x.x.x.174:5060;lr;ftag=1037646889;did=3dd.18dd4fd7>.
From: <sip:xxxx9 at x.x.x.174>;tag=1037646889.
To: <sip:xxxxxxx29858 at x.x.x.174>.
Call-ID: 1770373286-5062-8 at BJC.BGI.B.BFJ.
CSeq: 70 INVITE.
User-Agent: M01.
Content-Length: 0.
.


U x.x.x.166:5060 -> x.x.x.174:5060
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP x.x.x.174:5060;branch=z9hG4bK1bf1.f4e1b863.0.
Via: SIP/2.0/UDP x.x.x.47:2313;received=x.x.x.47;branch=z9hG4bK1606798992;rport=2313.
From: <sip:xxxx9 at x.x.x.174>;tag=1037646889.
To: <sip:xxxxxxx29858 at x.x.x.174>;tag=reUey9taa7r0N.
Call-ID: 1770373286-5062-8 at BJC.BGI.B.BFJ.
CSeq: 70 INVITE.
User-Agent: M01.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: path, replaces.
Allow-Events: talk, hold, conference, refer.
Proxy-Authenticate: Digest realm="x.x.x.174", nonce="fd658a68-d16e-11e5-b0fa-b321f35ec85c", algorithm=MD5, qop="auth".
Content-Length: 0.
.


U x.x.x.174:5060 -> x.x.x.166:5060
ACK sip:xxxxxxx29858 at x.x.x.166:5060 SIP/2.0.
Via: SIP/2.0/UDP x.x.x.174:5060;branch=z9hG4bK1bf1.f4e1b863.0.
From: <sip:xxxx9 at x.x.x.174>;tag=1037646889.
Call-ID: 1770373286-5062-8 at BJC.BGI.B.BFJ.
To: <sip:xxxxxxx29858 at x.x.x.174>;tag=reUey9taa7r0N.
CSeq: 70 ACK.
Max-Forwards: 70.
Content-Length: 0.
.


U x.x.x.174:5060 -> x.x.x.166:5060
INVITE sip:xxxxxxx29858 at x.x.x.166:5060 SIP/2.0.
Record-Route: <sip:x.x.x.174:5060;lr;ftag=1037646889;did=3dd.28dd4fd7>.
Via: SIP/2.0/UDP x.x.x.174:5060;branch=z9hG4bK2bf1.30ee31f1.0.
Via: SIP/2.0/UDP x.x.x.47:2313;received=x.x.x.47;branch=z9hG4bK1418332271;rport=2313.
From: <sip:xxxx9 at x.x.x.174>;tag=1037646889.
To: <sip:xxxxxxx29858 at x.x.x.174>.
Call-ID: 1770373286-5062-8 at BJC.BGI.B.BFJ.
CSeq: 71 INVITE.
Contact: <sip:xxxx9 at x.x.x.47:2313>.
Proxy-Authorization: Digest username="xxxx9", realm="x.x.x.174", nonce="fd658a68-d16e-11e5-b0fa-b321f35ec85c", uri="sip:xxxxxxx29858 at x.x.x.174", response="97445fab50fed9e6930856844ec41daf", algorithm=MD5, cnonce="05356127", qop=auth, nc=0000000b.
Max-Forwards: 30.
User-Agent: Grandstream GXP1400 1.0.6.11.
Privacy: none.
P-Preferred-Identity: <sip:xxxx9 at x.x.x.174>.
Supported: replaces, path, timer.
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE.
Content-Type: application/sdp.
Accept: application/sdp, application/dtmf-relay.
Content-Length:   285.
X-AUTH-IP: x.x.x.47.
.
v=0.
o=xxxx9 8001 8000 IN IP4 x.x.x.47.
s=SIP Call.
c=IN IP4 x.x.x.47.
t=0 0.
m=audio 5008 RTP/AVP 0 8 18 101.
a=sendrecv.
a=rtpmap:0 PCMU/8000.
a=ptime:20.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.


U x.x.x.166:5060 -> x.x.x.174:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP x.x.x.174:5060;branch=z9hG4bK2bf1.30ee31f1.0.
Via: SIP/2.0/UDP x.x.x.47:2313;received=x.x.x.47;branch=z9hG4bK1418332271;rport=2313.
Record-Route: <sip:x.x.x.174:5060;lr;ftag=1037646889;did=3dd.28dd4fd7>.
From: <sip:xxxx9 at x.x.x.174>;tag=1037646889.
To: <sip:xxxxxxx29858 at x.x.x.174>.
Call-ID: 1770373286-5062-8 at BJC.BGI.B.BFJ.
CSeq: 71 INVITE.
User-Agent: M01.
Content-Length: 0.
.


U x.x.x.166:5060 -> x.x.x.174:5060
INVITE sip:xxxxxxx29858 at x.x.x.3 SIP/2.0.
Via: SIP/2.0/UDP x.x.x.166;rport;branch=z9hG4bK789FQF04S4yte.
Max-Forwards: 29.
From: "xxxxxxx00181" <sip:xxxxxxx00181 at x.x.x.166>;tag=U96r3tDN11Urr.
To: <sip:xxxxxxx29858 at x.x.x.3>.
Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84.
CSeq: 87299453 INVITE.
Contact: <sip:gw+test_gateway at x.x.x.166:5060;transport=udp;gw=test_gateway>.
User-Agent: M01.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 324.
X-AUTH-IP: x.x.x.47.
Remote-Party-ID: "xxxxxxx00181" <sip:xxxxxxx00181 at x.x.x.166>;party=calling;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1455243062 1455243063 IN IP4 x.x.x.166.
s=FreeSWITCH.
c=IN IP4 x.x.x.166.
t=0 0.
m=audio 27972 RTP/AVP 8 0 9 3 18 101 13.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:9 G722/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.


U x.x.x.174:5060 -> x.x.x.166:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP x.x.x.166;received=x.x.x.166;rport=5060;branch=z9hG4bK789FQF04S4yte.
From: "xxxxxxx00181" <sip:xxxxxxx00181 at x.x.x.166>;tag=U96r3tDN11Urr.
To: <sip:xxxxxxx29858 at x.x.x.3>.
Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84.
CSeq: 87299453 INVITE.
Content-Length: 0.
Warning: 392 x.x.x.174:5060 "Noisy feedback tells:  pid=36603 req_src_ip=x.x.x.166 req_src_port=5060 in_uri=sip:xxxxxxx29858 at x.x.x.3 out_uri=sip:xxxxxxx29858 at x.x.x.3 via_cnt==1".
.


U x.x.x.3:5060 -> x.x.x.166:5060
INVITE sip:xxxxxxx29858 at x.x.x.166 SIP/2.0.
Record-Route: <sip:x.x.x.3:5060;lr;ftag=9pp9134ya08pe;did=3b1.30cc623>.
Via: SIP/2.0/UDP x.x.x.3:5060;branch=z9hG4bK93e3.c07e36f3.0.
Max-Forwards: 25.
From: "xxxxxxx00181" <sip:xxxxxxx00181 at x.x.x.6>;tag=9pp9134ya08pe.
To: <sip:xxxxxxx29858 at x.x.x.166>.
Call-ID: 0db3670b-4c12-1234-51b7-021923026132.
CSeq: 87299500 INVITE.
Contact: <sip:gw+SBC_gateway at x.x.x.6:5060;transport=udp;gw=SBC_gateway>.
User-Agent: M01.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Disposition: session.
X-AUTH-IP: x.x.x.174.
P-Accountcode: <null>.
X-FS-Support: update_display,send_info.
Remote-Party-ID: "xxxxxxx00181" <sip:xxxxxxx00181 at x.x.x.6>;party=calling;screen=yes;privacy=off.
Content-Type: application/sdp.
Content-Length: 261.
.
v=0.
o=FreeSWITCH 1455243574 1455243575 IN IP4 x.x.x.6.
s=FreeSWITCH.
c=IN IP4 x.x.x.6.
t=0 0.
m=audio 27554 RTP/AVP 8 0 9 3 98 18 101 13.
a=rtpmap:98 G7221/16000.
a=fmtp:98 bitrate=32000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.


U x.x.x.166:5060 -> x.x.x.3:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP x.x.x.3:5060;branch=z9hG4bK93e3.c07e36f3.0.
Record-Route: <sip:x.x.x.3:5060;lr;ftag=9pp9134ya08pe;did=3b1.30cc623>.
From: "xxxxxxx00181" <sip:xxxxxxx00181 at x.x.x.6>;tag=9pp9134ya08pe.
To: <sip:xxxxxxx29858 at x.x.x.166>.
Call-ID: 0db3670b-4c12-1234-51b7-021923026132.
CSeq: 87299500 INVITE.
User-Agent: M01.
Content-Length: 0.
.


U x.x.x.166:5060 -> x.x.x.3:5060
SIP/2.0 603 Decline.
Via: SIP/2.0/UDP x.x.x.3:5060;branch=z9hG4bK93e3.c07e36f3.0.
Max-Forwards: 25.
From: "xxxxxxx00181" <sip:xxxxxxx00181 at x.x.x.6>;tag=9pp9134ya08pe.
To: <sip:xxxxxxx29858 at x.x.x.166>;tag=vj0H5NyryajBm.
Call-ID: 0db3670b-4c12-1234-51b7-021923026132.
CSeq: 87299500 INVITE.
User-Agent: M01.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: path, replaces.
Allow-Events: talk, hold, conference, refer.
Reason: Q.850;cause=21;text="CALL_REJECTED".
Content-Length: 0.
.


U x.x.x.3:5060 -> x.x.x.166:5060
ACK sip:xxxxxxx29858 at x.x.x.166 SIP/2.0.
Via: SIP/2.0/UDP x.x.x.3:5060;branch=z9hG4bK93e3.c07e36f3.0.
From: "xxxxxxx00181" <sip:xxxxxxx00181 at x.x.x.6>;tag=9pp9134ya08pe.
Call-ID: 0db3670b-4c12-1234-51b7-021923026132.
To: <sip:xxxxxxx29858 at x.x.x.166>;tag=vj0H5NyryajBm.
CSeq: 87299500 ACK.
Max-Forwards: 70.
Content-Length: 0.
.


U x.x.x.174:5060 -> x.x.x.166:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP x.x.x.166;rport=5060;received=x.x.x.166;branch=z9hG4bK789FQF04S4yte.
Record-Route: <sip:x.x.x.3:5060;lr;ftag=U96r3tDN11Urr;did=58e.0e323136>.
Record-Route: <sip:x.x.x.174:5060;lr;ftag=U96r3tDN11Urr;did=58e.ed4c9095>.
From: "xxxxxxx00181" <sip:xxxxxxx00181 at x.x.x.166>;tag=U96r3tDN11Urr.
To: <sip:xxxxxxx29858 at x.x.x.3>;tag=tFUN7D89Nttyg.
Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84.
CSeq: 87299453 INVITE.
Contact: <sip:xxxxxxx29858 at x.x.x.3:5060;transport=udp>.
User-Agent: M01.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Disposition: session.
Remote-Party-ID: "xxxxxxx29858" <sip:xxxxxxx29858 at x.x.x.3>;party=calling;privacy=off;screen=no.
Content-Type: application/sdp.
Content-Length: 242.
.
v=0.
o=FreeSWITCH 1455252976 1455252977 IN IP4 x.x.x.6.
s=FreeSWITCH.
c=IN IP4 x.x.x.6.
t=0 0.
m=audio 18154 RTP/AVP 8 101 13.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.


U x.x.x.166:5060 -> x.x.x.174:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP x.x.x.174:5060;branch=z9hG4bK2bf1.30ee31f1.0.
Via: SIP/2.0/UDP x.x.x.47:2313;received=x.x.x.47;branch=z9hG4bK1418332271;rport=2313.
Record-Route: <sip:x.x.x.174:5060;lr;ftag=1037646889;did=3dd.28dd4fd7>.
From: <sip:xxxx9 at x.x.x.174>;tag=1037646889.
To: <sip:xxxxxxx29858 at x.x.x.174>;tag=t0D01ZvH4r55c.
Call-ID: 1770373286-5062-8 at BJC.BGI.B.BFJ.
CSeq: 71 INVITE.
Contact: <sip:xxxxxxx29858 at x.x.x.166:5060;transport=udp>.
User-Agent: M01.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 224.
.
v=0.
o=FreeSWITCH 1455243571 1455243572 IN IP4 x.x.x.166.
s=FreeSWITCH.
c=IN IP4 x.x.x.166.
t=0 0.
m=audio 27468 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.


U x.x.x.174:5060 -> x.x.x.166:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP x.x.x.166;rport=5060;received=x.x.x.166;branch=z9hG4bK789FQF04S4yte.
Record-Route: <sip:x.x.x.3:5060;lr;ftag=U96r3tDN11Urr;did=58e.0e323136>.
Record-Route: <sip:x.x.x.174:5060;lr;ftag=U96r3tDN11Urr;did=58e.ed4c9095>.
From: "xxxxxxx00181" <sip:xxxxxxx00181 at x.x.x.166>;tag=U96r3tDN11Urr.
To: <sip:xxxxxxx29858 at x.x.x.3>;tag=tFUN7D89Nttyg.
Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84.
CSeq: 87299453 INVITE.
Contact: <sip:xxxxxxx29858 at x.x.x.3:5060;transport=udp>.
User-Agent: M01.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Disposition: session.
Remote-Party-ID: "Outbound Call" <sip:+xxxxxxx29858 at x.x.x.3>;party=calling;privacy=off;screen=no.
Content-Type: application/sdp.
Content-Length: 242.
.
v=0.
o=FreeSWITCH 1455252976 1455252977 IN IP4 x.x.x.6.
s=FreeSWITCH.
c=IN IP4 x.x.x.6.
t=0 0.
m=audio 18154 RTP/AVP 8 101 13.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.


U x.x.x.166:5060 -> x.x.x.174:5060
ACK sip:xxxxxxx29858 at x.x.x.3:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP x.x.x.166;rport;branch=z9hG4bK8H38raH8pDNDa.
Route: <sip:x.x.x.174:5060;lr;ftag=U96r3tDN11Urr;did=58e.ed4c9095>.
Route: <sip:x.x.x.3:5060;lr;ftag=U96r3tDN11Urr;did=58e.0e323136>.
Max-Forwards: 70.
From: "xxxxxxx00181" <sip:xxxxxxx00181 at x.x.x.166>;tag=U96r3tDN11Urr.
To: <sip:xxxxxxx29858 at x.x.x.3>;tag=tFUN7D89Nttyg.
Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84.
CSeq: 87299453 ACK.
Contact: <sip:gw+test_gateway at x.x.x.166:5060;transport=udp;gw=test_gateway>.
Content-Length: 0.
.


U x.x.x.166:5060 -> x.x.x.174:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP x.x.x.174:5060;branch=z9hG4bK2bf1.30ee31f1.0.
Via: SIP/2.0/UDP x.x.x.47:2313;received=x.x.x.47;branch=z9hG4bK1418332271;rport=2313.
Record-Route: <sip:x.x.x.174:5060;lr;ftag=1037646889;did=3dd.28dd4fd7>.
From: <sip:xxxx9 at x.x.x.174>;tag=1037646889.
To: <sip:xxxxxxx29858 at x.x.x.174>;tag=t0D01ZvH4r55c.
Call-ID: 1770373286-5062-8 at BJC.BGI.B.BFJ.
CSeq: 71 INVITE.
Contact: <sip:xxxxxxx29858 at x.x.x.166:5060;transport=udp>.
User-Agent: M01.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 224.
.
v=0.
o=FreeSWITCH 1455243571 1455243572 IN IP4 x.x.x.166.
s=FreeSWITCH.
c=IN IP4 x.x.x.166.
t=0 0.
m=audio 27468 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.


U x.x.x.174:5060 -> x.x.x.166:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP x.x.x.166;rport=5060;received=x.x.x.166;branch=z9hG4bK8H38raH8pDNDa.
Record-Route: <sip:x.x.x.3:5060;lr;ftag=U96r3tDN11Urr;did=58e.0e323136>.
Record-Route: <sip:x.x.x.174:5060;lr;ftag=U96r3tDN11Urr;did=58e.ed4c9095>.
From: "xxxxxxx00181" <sip:xxxxxxx00181 at x.x.x.166>;tag=U96r3tDN11Urr.
To: <sip:xxxxxxx29858 at x.x.x.3>;tag=tFUN7D89Nttyg.
Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84.
CSeq: 87299453 INVITE.
Contact: <sip:xxxxxxx29858 at x.x.x.3:5060;transport=udp>.
User-Agent: M01.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Disposition: session.
Remote-Party-ID: "Outbound Call" <sip:+xxxxxxx29858 at x.x.x.3>;party=calling;privacy=off;screen=no.
Content-Type: application/sdp.
Content-Length: 242.
.
v=0.
o=FreeSWITCH 1455252976 1455252977 IN IP4 x.x.x.6.
s=FreeSWITCH.
c=IN IP4 x.x.x.6.
t=0 0.
m=audio 18154 RTP/AVP 8 101 13.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.


U x.x.x.166:5060 -> x.x.x.174:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP x.x.x.174:5060;branch=z9hG4bK2bf1.30ee31f1.0.
Via: SIP/2.0/UDP x.x.x.47:2313;received=x.x.x.47;branch=z9hG4bK1418332271;rport=2313.
Record-Route: <sip:x.x.x.174:5060;lr;ftag=1037646889;did=3dd.28dd4fd7>.
From: <sip:xxxx9 at x.x.x.174>;tag=1037646889.
To: <sip:xxxxxxx29858 at x.x.x.174>;tag=t0D01ZvH4r55c.
Call-ID: 1770373286-5062-8 at BJC.BGI.B.BFJ.
CSeq: 71 INVITE.
Contact: <sip:xxxxxxx29858 at x.x.x.166:5060;transport=udp>.
User-Agent: M01.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 224.
.
v=0.
o=FreeSWITCH 1455243571 1455243572 IN IP4 x.x.x.166.
s=FreeSWITCH.
c=IN IP4 x.x.x.166.
t=0 0.
m=audio 27468 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.


U x.x.x.174:5060 -> x.x.x.166:5060
ACK sip:xxxxxxx29858 at x.x.x.166:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP x.x.x.174:5060;branch=z9hG4bK2bf1.30ee31f1.2.
Via: SIP/2.0/UDP x.x.x.47:2313;received=x.x.x.47;branch=z9hG4bK1857959530;rport=2313.
From: <sip:xxxx9 at x.x.x.174>;tag=1037646889.
To: <sip:xxxxxxx29858 at x.x.x.174>;tag=t0D01ZvH4r55c.
Call-ID: 1770373286-5062-8 at BJC.BGI.B.BFJ.
CSeq: 71 ACK.
Contact: <sip:xxxx9 at x.x.x.47:2313>.
Proxy-Authorization: Digest username="xxxx9", realm="x.x.x.174", nonce="fd658a68-d16e-11e5-b0fa-b321f35ec85c", uri="sip:xxxxxxx29858 at x.x.x.174", response="97445fab50fed9e6930856844ec41daf", algorithm=MD5, cnonce="05356127", qop=auth, nc=0000000b.
Max-Forwards: 30.
Supported: replaces, path, timer.
User-Agent: Grandstream GXP1400 1.0.6.11.
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE.
Content-Length: 0.
P-hint: rr-enforced.
X-AUTH-IP: x.x.x.47.
.


U x.x.x.174:5060 -> x.x.x.166:5060
ACK sip:xxxxxxx29858 at x.x.x.166:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP x.x.x.174:5060;branch=z9hG4bK2bf1.30ee31f1.2.
Via: SIP/2.0/UDP x.x.x.47:2313;received=x.x.x.47;branch=z9hG4bK1857959530;rport=2313.
From: <sip:xxxx9 at x.x.x.174>;tag=1037646889.
To: <sip:xxxxxxx29858 at x.x.x.174>;tag=t0D01ZvH4r55c.
Call-ID: 1770373286-5062-8 at BJC.BGI.B.BFJ.
CSeq: 71 ACK.
Contact: <sip:xxxx9 at x.x.x.47:2313>.
Proxy-Authorization: Digest username="xxxx9", realm="x.x.x.174", nonce="fd658a68-d16e-11e5-b0fa-b321f35ec85c", uri="sip:xxxxxxx29858 at x.x.x.174", response="97445fab50fed9e6930856844ec41daf", algorithm=MD5, cnonce="05356127", qop=auth, nc=0000000b.
Max-Forwards: 30.
Supported: replaces, path, timer.
User-Agent: Grandstream GXP1400 1.0.6.11.
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE.
Content-Length: 0.
P-hint: rr-enforced.
X-AUTH-IP: x.x.x.47.
.


U x.x.x.174:5060 -> x.x.x.166:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP x.x.x.166;rport=5060;received=x.x.x.166;branch=z9hG4bK8H38raH8pDNDa.
Record-Route: <sip:x.x.x.3:5060;lr;ftag=U96r3tDN11Urr;did=58e.0e323136>.
Record-Route: <sip:x.x.x.174:5060;lr;ftag=U96r3tDN11Urr;did=58e.ed4c9095>.
From: "xxxxxxx00181" <sip:xxxxxxx00181 at x.x.x.166>;tag=U96r3tDN11Urr.
To: <sip:xxxxxxx29858 at x.x.x.3>;tag=tFUN7D89Nttyg.
Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84.
CSeq: 87299453 INVITE.
Contact: <sip:xxxxxxx29858 at x.x.x.3:5060;transport=udp>.
User-Agent: M01.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Disposition: session.
Remote-Party-ID: "Outbound Call" <sip:+xxxxxxx29858 at x.x.x.3>;party=calling;privacy=off;screen=no.
Content-Type: application/sdp.
Content-Length: 242.
.
v=0.
o=FreeSWITCH 1455252976 1455252977 IN IP4 x.x.x.6.
s=FreeSWITCH.
c=IN IP4 x.x.x.6.
t=0 0.
m=audio 18154 RTP/AVP 8 101 13.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.


U x.x.x.174:5060 -> x.x.x.166:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP x.x.x.166;rport=5060;received=x.x.x.166;branch=z9hG4bK8H38raH8pDNDa.
Record-Route: <sip:x.x.x.3:5060;lr;ftag=U96r3tDN11Urr;did=58e.0e323136>.
Record-Route: <sip:x.x.x.174:5060;lr;ftag=U96r3tDN11Urr;did=58e.ed4c9095>.
From: "xxxxxxx00181" <sip:xxxxxxx00181 at x.x.x.166>;tag=U96r3tDN11Urr.
To: <sip:xxxxxxx29858 at x.x.x.3>;tag=tFUN7D89Nttyg.
Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84.
CSeq: 87299453 INVITE.
Contact: <sip:xxxxxxx29858 at x.x.x.3:5060;transport=udp>.
User-Agent: M01.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Disposition: session.
Remote-Party-ID: "Outbound Call" <sip:+xxxxxxx29858 at x.x.x.3>;party=calling;privacy=off;screen=no.
Content-Type: application/sdp.
Content-Length: 242.
.
v=0.
o=FreeSWITCH 1455252976 1455252977 IN IP4 x.x.x.6.
s=FreeSWITCH.
c=IN IP4 x.x.x.6.
t=0 0.
m=audio 18154 RTP/AVP 8 101 13.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.


U x.x.x.174:5060 -> x.x.x.166:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP x.x.x.166;rport=5060;received=x.x.x.166;branch=z9hG4bK8H38raH8pDNDa.
Record-Route: <sip:x.x.x.3:5060;lr;ftag=U96r3tDN11Urr;did=58e.0e323136>.
Record-Route: <sip:x.x.x.174:5060;lr;ftag=U96r3tDN11Urr;did=58e.ed4c9095>.
From: "xxxxxxx00181" <sip:xxxxxxx00181 at x.x.x.166>;tag=U96r3tDN11Urr.
To: <sip:xxxxxxx29858 at x.x.x.3>;tag=tFUN7D89Nttyg.
Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84.
CSeq: 87299453 INVITE.
Contact: <sip:xxxxxxx29858 at x.x.x.6:5060;transport=udp>.
User-Agent: M01.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Disposition: session.
Remote-Party-ID: "Outbound Call" <sip:+xxxxxxx29858 at x.x.x.3>;party=calling;privacy=off;screen=no.
Content-Type: application/sdp.
Content-Length: 242.
.
v=0.
o=FreeSWITCH 1455252976 1455252977 IN IP4 x.x.x.6.
s=FreeSWITCH.
c=IN IP4 x.x.x.6.
t=0 0.
m=audio 18154 RTP/AVP 8 101 13.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.


U x.x.x.174:5060 -> x.x.x.166:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP x.x.x.166;rport=5060;received=x.x.x.166;branch=z9hG4bK8H38raH8pDNDa.
Record-Route: <sip:x.x.x.3:5060;lr;ftag=U96r3tDN11Urr;did=58e.0e323136>.
Record-Route: <sip:x.x.x.174:5060;lr;ftag=U96r3tDN11Urr;did=58e.ed4c9095>.
From: "xxxxxxx00181" <sip:xxxxxxx00181 at x.x.x.166>;tag=U96r3tDN11Urr.
To: <sip:xxxxxxx29858 at x.x.x.3>;tag=tFUN7D89Nttyg.
Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84.
CSeq: 87299453 INVITE.
Contact: <sip:xxxxxxx29858 at x.x.x.6:5060;transport=udp>.
User-Agent: M01.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Disposition: session.
Remote-Party-ID: "Outbound Call" <sip:+xxxxxxx29858 at x.x.x.3>;party=calling;privacy=off;screen=no.
Content-Type: application/sdp.
Content-Length: 242.
.
v=0.
o=FreeSWITCH 1455252976 1455252977 IN IP4 x.x.x.6.
s=FreeSWITCH.
c=IN IP4 x.x.x.6.
t=0 0.
m=audio 18154 RTP/AVP 8 101 13.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.


U x.x.x.174:5060 -> x.x.x.166:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP x.x.x.166;rport=5060;received=x.x.x.166;branch=z9hG4bK8H38raH8pDNDa.
Record-Route: <sip:x.x.x.3:5060;lr;ftag=U96r3tDN11Urr;did=58e.0e323136>.
Record-Route: <sip:x.x.x.174:5060;lr;ftag=U96r3tDN11Urr;did=58e.ed4c9095>.
From: "xxxxxxx00181" <sip:xxxxxxx00181 at x.x.x.166>;tag=U96r3tDN11Urr.
To: <sip:xxxxxxx29858 at x.x.x.3>;tag=tFUN7D89Nttyg.
Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84.
CSeq: 87299453 INVITE.
Contact: <sip:xxxxxxx29858 at x.x.x.6:5060;transport=udp>.
User-Agent: M01.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Disposition: session.
Remote-Party-ID: "Outbound Call" <sip:+xxxxxxx29858 at x.x.x.3>;party=calling;privacy=off;screen=no.
Content-Type: application/sdp.
Content-Length: 242.
.
v=0.
o=FreeSWITCH 1455252976 1455252977 IN IP4 x.x.x.6.
s=FreeSWITCH.
c=IN IP4 x.x.x.6.
t=0 0.
m=audio 18154 RTP/AVP 8 101 13.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.


U x.x.x.174:5060 -> x.x.x.166:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP x.x.x.166;rport=5060;received=x.x.x.166;branch=z9hG4bK8H38raH8pDNDa.
Record-Route: <sip:x.x.x.3:5060;lr;ftag=U96r3tDN11Urr;did=58e.0e323136>.
Record-Route: <sip:x.x.x.174:5060;lr;ftag=U96r3tDN11Urr;did=58e.ed4c9095>.
From: "xxxxxxx00181" <sip:xxxxxxx00181 at x.x.x.166>;tag=U96r3tDN11Urr.
To: <sip:xxxxxxx29858 at x.x.x.3>;tag=tFUN7D89Nttyg.
Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84.
CSeq: 87299453 INVITE.
Contact: <sip:xxxxxxx29858 at x.x.x.6:5060;transport=udp>.
User-Agent: M01.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Disposition: session.
Remote-Party-ID: "Outbound Call" <sip:+xxxxxxx29858 at x.x.x.3>;party=calling;privacy=off;screen=no.
Content-Type: application/sdp.
Content-Length: 242.
.
v=0.
o=FreeSWITCH 1455252976 1455252977 IN IP4 x.x.x.6.
s=FreeSWITCH.
c=IN IP4 x.x.x.6.
t=0 0.
m=audio 18154 RTP/AVP 8 101 13.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.


U x.x.x.174:5060 -> x.x.x.166:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP x.x.x.166;rport=5060;received=x.x.x.166;branch=z9hG4bK8H38raH8pDNDa.
Record-Route: <sip:x.x.x.3:5060;lr;ftag=U96r3tDN11Urr;did=58e.0e323136>.
Record-Route: <sip:x.x.x.174:5060;lr;ftag=U96r3tDN11Urr;did=58e.ed4c9095>.
From: "xxxxxxx00181" <sip:xxxxxxx00181 at x.x.x.166>;tag=U96r3tDN11Urr.
To: <sip:xxxxxxx29858 at x.x.x.3>;tag=tFUN7D89Nttyg.
Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84.
CSeq: 87299453 INVITE.
Contact: <sip:xxxxxxx29858 at x.x.x.6:5060;transport=udp>.
User-Agent: M01.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Disposition: session.
Remote-Party-ID: "Outbound Call" <sip:+xxxxxxx29858 at x.x.x.3>;party=calling;privacy=off;screen=no.
Content-Type: application/sdp.
Content-Length: 242.
.
v=0.
o=FreeSWITCH 1455252976 1455252977 IN IP4 x.x.x.6.
s=FreeSWITCH.
c=IN IP4 x.x.x.6.
t=0 0.
m=audio 18154 RTP/AVP 8 101 13.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.


U x.x.x.174:5060 -> x.x.x.166:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP x.x.x.166;rport=5060;received=x.x.x.166;branch=z9hG4bK8H38raH8pDNDa.
Record-Route: <sip:x.x.x.3:5060;lr;ftag=U96r3tDN11Urr;did=58e.0e323136>.
Record-Route: <sip:x.x.x.174:5060;lr;ftag=U96r3tDN11Urr;did=58e.ed4c9095>.
From: "xxxxxxx00181" <sip:xxxxxxx00181 at x.x.x.166>;tag=U96r3tDN11Urr.
To: <sip:xxxxxxx29858 at x.x.x.3>;tag=tFUN7D89Nttyg.
Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84.
CSeq: 87299453 INVITE.
Contact: <sip:xxxxxxx29858 at x.x.x.6:5060;transport=udp>.
User-Agent: M01.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Disposition: session.
Remote-Party-ID: "Outbound Call" <sip:+xxxxxxx29858 at x.x.x.3>;party=calling;privacy=off;screen=no.
Content-Type: application/sdp.
Content-Length: 242.
.
v=0.
o=FreeSWITCH 1455252976 1455252977 IN IP4 x.x.x.6.
s=FreeSWITCH.
c=IN IP4 x.x.x.6.
t=0 0.
m=audio 18154 RTP/AVP 8 101 13.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.


U x.x.x.174:5060 -> x.x.x.166:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP x.x.x.166;rport=5060;received=x.x.x.166;branch=z9hG4bK8H38raH8pDNDa.
Record-Route: <sip:x.x.x.3:5060;lr;ftag=U96r3tDN11Urr;did=58e.0e323136>.
Record-Route: <sip:x.x.x.174:5060;lr;ftag=U96r3tDN11Urr;did=58e.ed4c9095>.
From: "xxxxxxx00181" <sip:xxxxxxx00181 at x.x.x.166>;tag=U96r3tDN11Urr.
To: <sip:xxxxxxx29858 at x.x.x.3>;tag=tFUN7D89Nttyg.
Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84.
CSeq: 87299453 INVITE.
Contact: <sip:xxxxxxx29858 at x.x.x.6:5060;transport=udp>.
User-Agent: M01.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Disposition: session.
Remote-Party-ID: "Outbound Call" <sip:+xxxxxxx29858 at x.x.x.3>;party=calling;privacy=off;screen=no.
Content-Type: application/sdp.
Content-Length: 242.
.
v=0.
o=FreeSWITCH 1455252976 1455252977 IN IP4 x.x.x.6.
s=FreeSWITCH.
c=IN IP4 x.x.x.6.
t=0 0.
m=audio 18154 RTP/AVP 8 101 13.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.


U x.x.x.174:5060 -> x.x.x.166:5060
BYE sip:gw+test_gateway at x.x.x.166:5060;transport=udp;gw=test_gateway SIP/2.0.
Via: SIP/2.0/UDP x.x.x.174:5060;branch=z9hG4bKb4be.671066d2.0.
Via: SIP/2.0/UDP x.x.x.3:5060;branch=z9hG4bKb4be.fba2e636.0.
Max-Forwards: 29.
From: <sip:xxxxxxx29858 at x.x.x.3>;tag=tFUN7D89Nttyg.
To: "xxxxxxx00181" <sip:xxxxxxx00181 at x.x.x.166>;tag=U96r3tDN11Urr.
Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84.
CSeq: 87299520 BYE.
Contact: <sip:xxxxxxx29858 at x.x.x.6:5060;transport=udp>.
User-Agent: M01.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: path, replaces.
Reason: SIP;cause=408;text="ACK Timeout".
P-hint: rr-enforced.
Content-Length: 0.
P-hint: rr-enforced.
X-AUTH-IP: x.x.x.3.
.


U x.x.x.166:5060 -> x.x.x.174:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP x.x.x.174:5060;branch=z9hG4bKb4be.671066d2.0.
Via: SIP/2.0/UDP x.x.x.3:5060;branch=z9hG4bKb4be.fba2e636.0.
From: <sip:xxxxxxx29858 at x.x.x.3>;tag=tFUN7D89Nttyg.
To: "xxxxxxx00181" <sip:xxxxxxx00181 at x.x.x.166>;tag=U96r3tDN11Urr.
Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84.
CSeq: 87299520 BYE.
User-Agent: M01.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: path, replaces.
Content-Length: 0.
.


U x.x.x.166:5060 -> x.x.x.174:5060
BYE sip:xxxx9 at x.x.x.47:2313 SIP/2.0.
Via: SIP/2.0/UDP x.x.x.166;rport;branch=z9hG4bK9tv1t51BmpB0N.
Route: <sip:x.x.x.174:5060;lr;ftag=1037646889;did=3dd.28dd4fd7>.
Max-Forwards: 70.
From: <sip:xxxxxxx29858 at x.x.x.174>;tag=t0D01ZvH4r55c.
To: <sip:xxxx9 at x.x.x.174>;tag=1037646889.
Call-ID: 1770373286-5062-8 at BJC.BGI.B.BFJ.
CSeq: 87299474 BYE.
User-Agent: M01.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: path, replaces.
Reason: SIP;cause=408;text="ACK Timeout".
Content-Length: 0.
P-hint: rr-enforced.
X-AUTH-IP: x.x.x.3.
.


U x.x.x.174:5060 -> x.x.x.166:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP x.x.x.166;received=x.x.x.166;rport=5060;branch=z9hG4bK9tv1t51BmpB0N.
From: <sip:xxxxxxx29858 at x.x.x.174>;tag=t0D01ZvH4r55c.
To: <sip:xxxx9 at x.x.x.174>;tag=1037646889.
Call-ID: 1770373286-5062-8 at BJC.BGI.B.BFJ.
CSeq: 87299474 BYE.
Contact: <sip:xxxx9 at x.x.x.47:2313>.
Supported: replaces, path, timer.
User-Agent: Grandstream GXP1400 1.0.6.11.
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE.
Content-Length: 0.
.

From avi at avimarcus.net  Mon Feb 22 09:36:14 2016
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 22 Feb 2016 06:36:14 +0000
Subject: [Freeswitch-users] Call dropping after 32 seconds
In-Reply-To: <CAPQet1zs08bGNnGDVmsLH3R3VbvMCJW=8Cx3U+KPxwGpYSuzAA@mail.gmail.com>
References: <CAPQet1zs08bGNnGDVmsLH3R3VbvMCJW=8Cx3U+KPxwGpYSuzAA@mail.gmail.com>
Message-ID: <0000015307b150a5-132a41d6-5bea-4af1-9686-2653df68a3b6-000000@email.amazonses.com>

5 second response: 32 seconds is a timer/[network/NAT] issue.

You have lots of 200s to the user since it's waiting for an ACK and keeps
retrying, but for whatever network reason (router... sip alg?), it isn't
getting one, so it triggers a timer to stop the call.


-Avi Marcus
BestFone

On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel <rutu.patel at inextrix.com> wrote:

> Hello,
>
> Having issue of call dropping after 32 seconds, here are the details-
>
> x.x.x.174: opensips server
> x.x.x.166: freeswitch server
> x.x.x.3:     another opensips server which is registered as gateway on
> above freeswitch server
> x.x.x.6: freeswitch server
> x.x.x.47:  server through which the user is registered
> I am trying to call from xxxx9 to xxxxxxx29858
> xxxxxxx00181 is caller-id name and caller-id number
>
> Call flow is like this:
> registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips
> server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server)
>
> Outbound-proxy is set to x.x.x.174 in Gateway configuration.
>
> 1) Call is initiated by the user(xxxx9) registered with host x.x.x.174
> 2) Call hit the freeswitch server x.x.x.166
> 3) After '180 Ringing' and '183 Session Progress' packet sending-receiving
> started between 'x.x.x.174' and 'x.x.x.166' through the gateway x.x.x.3
> But after 32 seconds call is dropped,
> Within 32 seconds audio is ok from both end so it should not be the RTP
> issue.
> Here I have attached the file with sip logs, you can observer from the
> file that, there are many '200 OK' from x.x.x.174 to x.x.x.166
> and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then
> 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped.
>
> What is wrong here? Any help would be appreciated here.
>
> Here is the file with sip logs
> --
> Thanks,
> Rutu Patel
> <http://www.inextrix.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 gascagonzalo at gmail.com  Mon Feb 22 12:58:12 2016
From: gascagonzalo at gmail.com (Gonzalo Gasca Meza)
Date: Mon, 22 Feb 2016 01:58:12 -0800
Subject: [Freeswitch-users] Freeswitch libvpx2-dev compile error Debian 8
Message-ID: <CAL+DTgLM8q8qhpgVOzq_asfiGMn7Ub=1CQ1Hf3ztnPiyBPee3g@mail.gmail.com>

Compiling latest FS version getting the following error when running make:

You must install libvpx2-dev to build mod_av.  Stop.

*Using this guide:*

https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video

*Version: **Debian 8.3 x64*

*Troubleshooting*

Previous version (Last week) was working fine in 2 different systems.

*git log*

commit 3bd26eaa6b2dff1d80cac4e58fc860fb96f92935

Author: Anthony Minessale <anthm at freeswitch.org>

Date:   Fri Feb 19 09:34:38 2016 -0600

    FS-8847 #resolve [Silence Error on shutdown of video call]

commit 9f4f67df5d2d9606ba88023b1bd05d7b88c69dad

Author: Anthony Minessale <anthm at freeswitch.org>

Date:   Thu Feb 18 17:04:01 2016 -0600

*git diff*

https://freeswitch.org/stash/projects/FS/repos/freeswitch/diff/src/mod/applications/mod_av/Makefile.am?until=4ad0aa91a4a34fd10782dfe21aba74d49724f995


*logs (make)*


making all mod_amr

make[4]: Entering directory '/usr/src/freeswitch.git/src/mod/codecs/mod_amr'

make  all-am

make[5]: Entering directory '/usr/src/freeswitch.git/src/mod/codecs/mod_amr'

  CC       mod_amr_la-mod_amr.lo

  CCLD     mod_amr.la

make[5]: Leaving directory '/usr/src/freeswitch.git/src/mod/codecs/mod_amr'

make[4]: Leaving directory '/usr/src/freeswitch.git/src/mod/codecs/mod_amr'

making all mod_av

make[4]: Entering directory
'/usr/src/freeswitch.git/src/mod/applications/mod_av'

Makefile:915: *** You must install libvpx2-dev to build mod_av.  Stop.

make[4]: Leaving directory
'/usr/src/freeswitch.git/src/mod/applications/mod_av'

Makefile:648: recipe for target 'mod_av-all' failed


/usr/src/freeswitch.git/src/mod/applications/mod_av# tree

.

??? avcodec.c

??? avformat.c

??? Makefile

??? Makefile.am

??? Makefile.in

??? mod_av.c


0 directories, 6 files

root at nkn-freeswitch-sfo1:/usr/src/freeswitch.git/src/mod/applications/mod_av#
ls -alh

total 208K

drwxr-xr-x  3 root root 4.0K Feb 22 04:46 .

drwxr-xr-x 59 root root 4.0K Feb 22 04:20 ..

-rw-r--r--  1 root root  46K Feb 22 04:20 avcodec.c

-rw-r--r--  1 root root  63K Feb 22 04:20 avformat.c

drwxr-xr-x  2 root root 4.0K Feb 22 04:24 .deps

-rw-r--r--  1 root root  35K Feb 22 04:24 Makefile

-rw-r--r--  1 root root  828 Feb 22 04:20 Makefile.am

-rw-r--r--  1 root root  33K Feb 22 04:23 Makefile.in

-rw-r--r--  1 root root 4.3K Feb 22 04:20 mod_av.c

*Found*

http://lists.freeswitch.org/pipermail/freeswitch-users/2015-October/116401.html
But not sure how to proceed for this release

Thanks
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From gmaruzz at gmail.com  Mon Feb 22 14:12:28 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 22 Feb 2016 12:12:28 +0100
Subject: [Freeswitch-users] Freeswitch libvpx2-dev compile error Debian 8
In-Reply-To: <CAL+DTgLM8q8qhpgVOzq_asfiGMn7Ub=1CQ1Hf3ztnPiyBPee3g@mail.gmail.com>
References: <CAL+DTgLM8q8qhpgVOzq_asfiGMn7Ub=1CQ1Hf3ztnPiyBPee3g@mail.gmail.com>
Message-ID: <CALXCt0qKs89MrSvGk7ckx9M+geTdX+BRzW5AniMVh+uJ4ht6kg@mail.gmail.com>

As told in irc, you must install *unstable* repository for compiling latest
git master.

Check on www.freeswitch.conf/confluence the page for debian 8 jessie with
updated instruction
Il 22/Feb/2016 10:59, "Gonzalo Gasca Meza" <gascagonzalo at gmail.com> ha
scritto:

> Compiling latest FS version getting the following error when running make:
>
> You must install libvpx2-dev to build mod_av.  Stop.
>
> *Using this guide:*
>
> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video
>
> *Version: **Debian 8.3 x64*
>
> *Troubleshooting*
>
> Previous version (Last week) was working fine in 2 different systems.
>
> *git log*
>
> commit 3bd26eaa6b2dff1d80cac4e58fc860fb96f92935
>
> Author: Anthony Minessale <anthm at freeswitch.org>
>
> Date:   Fri Feb 19 09:34:38 2016 -0600
>
>     FS-8847 #resolve [Silence Error on shutdown of video call]
>
> commit 9f4f67df5d2d9606ba88023b1bd05d7b88c69dad
>
> Author: Anthony Minessale <anthm at freeswitch.org>
>
> Date:   Thu Feb 18 17:04:01 2016 -0600
>
> *git diff*
>
>
> https://freeswitch.org/stash/projects/FS/repos/freeswitch/diff/src/mod/applications/mod_av/Makefile.am?until=4ad0aa91a4a34fd10782dfe21aba74d49724f995
>
>
> *logs (make)*
>
>
> making all mod_amr
>
> make[4]: Entering directory
> '/usr/src/freeswitch.git/src/mod/codecs/mod_amr'
>
> make  all-am
>
> make[5]: Entering directory
> '/usr/src/freeswitch.git/src/mod/codecs/mod_amr'
>
>   CC       mod_amr_la-mod_amr.lo
>
>   CCLD     mod_amr.la
>
> make[5]: Leaving directory '/usr/src/freeswitch.git/src/mod/codecs/mod_amr'
>
> make[4]: Leaving directory '/usr/src/freeswitch.git/src/mod/codecs/mod_amr'
>
> making all mod_av
>
> make[4]: Entering directory
> '/usr/src/freeswitch.git/src/mod/applications/mod_av'
>
> Makefile:915: *** You must install libvpx2-dev to build mod_av.  Stop.
>
> make[4]: Leaving directory
> '/usr/src/freeswitch.git/src/mod/applications/mod_av'
>
> Makefile:648: recipe for target 'mod_av-all' failed
>
>
> /usr/src/freeswitch.git/src/mod/applications/mod_av# tree
>
> .
>
> ??? avcodec.c
>
> ??? avformat.c
>
> ??? Makefile
>
> ??? Makefile.am
>
> ??? Makefile.in
>
> ??? mod_av.c
>
>
> 0 directories, 6 files
>
> root at nkn-freeswitch-sfo1:/usr/src/freeswitch.git/src/mod/applications/mod_av#
> ls -alh
>
> total 208K
>
> drwxr-xr-x  3 root root 4.0K Feb 22 04:46 .
>
> drwxr-xr-x 59 root root 4.0K Feb 22 04:20 ..
>
> -rw-r--r--  1 root root  46K Feb 22 04:20 avcodec.c
>
> -rw-r--r--  1 root root  63K Feb 22 04:20 avformat.c
>
> drwxr-xr-x  2 root root 4.0K Feb 22 04:24 .deps
>
> -rw-r--r--  1 root root  35K Feb 22 04:24 Makefile
>
> -rw-r--r--  1 root root  828 Feb 22 04:20 Makefile.am
>
> -rw-r--r--  1 root root  33K Feb 22 04:23 Makefile.in
>
> -rw-r--r--  1 root root 4.3K Feb 22 04:20 mod_av.c
>
> *Found*
>
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2015-October/116401.html
> But not sure how to proceed for this release
>
> 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 aubalde at presenceco.com  Mon Feb 22 18:51:13 2016
From: aubalde at presenceco.com (=?UTF-8?Q?Agust=C3=AD_Ubalde?=)
Date: Mon, 22 Feb 2016 16:51:13 +0100
Subject: [Freeswitch-users] Invalid sdp on WebRTC connections
Message-ID: <CAKThY06c7v=ftYa=qX93UHkUJUvTcPJq0B5pmH3SoDVfJFuWPw@mail.gmail.com>

We have a WebRTC environment with Freeswitch and sipml5. The fact is that
when a WebRTC extension receives a call, the SDP that sends Freeswitch not
show the correct candidate:
*======================================================================*
*22d33210-d709-11e5-bdcd-078a69e08f30 Local SDP:*
*22d33210-d709-11e5-bdcd-078a69e08f30 v=0*
*22d33210-d709-11e5-bdcd-078a69e08f30 o=FreeSWITCH 1455870972 1455870973 IN
IP4 X.X.X.X*
*22d33210-d709-11e5-bdcd-078a69e08f30 s=FreeSWITCH*
*22d33210-d709-11e5-bdcd-078a69e08f30 c=IN IP4 X.X.X.X*
*======================================================================*

Where X.X.X.X is the internal IP of Freeswitch beghind the NAT.
Any idea?


Thanks,

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

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From mgiammarco at gmail.com  Mon Feb 22 20:31:28 2016
From: mgiammarco at gmail.com (Mario Giammarco)
Date: Mon, 22 Feb 2016 17:31:28 +0000 (UTC)
Subject: [Freeswitch-users] Recording and mono directional webrtc video
Message-ID: <loom.20160222T182837-828@post.gmane.org>

Hello,
I am evaluating freeswitch for this purpose: I need a bidirectional audio 
call with one direction video only from caller. And I need to record audio 
and video from caller.
I need to use webrtc protocol.
I have seen old threads that show that FreeSwitch has support for these 
things but I would like to be sure that the implementation is now on main 
tree and it is really supported.

Thanks,
Mario



From anthony.minessale at gmail.com  Mon Feb 22 22:15:48 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Feb 2016 13:15:48 -0600
Subject: [Freeswitch-users] Invalid sdp on WebRTC connections
In-Reply-To: <CAKThY06c7v=ftYa=qX93UHkUJUvTcPJq0B5pmH3SoDVfJFuWPw@mail.gmail.com>
References: <CAKThY06c7v=ftYa=qX93UHkUJUvTcPJq0B5pmH3SoDVfJFuWPw@mail.gmail.com>
Message-ID: <CAKbxfG-0WAemipjMZHb96HvV9=NDa+-pVRVWMeteSy-G5RD1hw@mail.gmail.com>

See

ext-rtp-ip and

apply-candidate-acl



On Mon, Feb 22, 2016 at 9:51 AM, Agust? Ubalde <aubalde at presenceco.com>
wrote:

> We have a WebRTC environment with Freeswitch and sipml5. The fact is that
> when a WebRTC extension receives a call, the SDP that sends Freeswitch not
> show the correct candidate:
> *======================================================================*
> *22d33210-d709-11e5-bdcd-078a69e08f30 Local SDP:*
> *22d33210-d709-11e5-bdcd-078a69e08f30 v=0*
> *22d33210-d709-11e5-bdcd-078a69e08f30 o=FreeSWITCH 1455870972 1455870973
> IN IP4 X.X.X.X*
> *22d33210-d709-11e5-bdcd-078a69e08f30 s=FreeSWITCH*
> *22d33210-d709-11e5-bdcd-078a69e08f30 c=IN IP4 X.X.X.X*
> *======================================================================*
>
> Where X.X.X.X is the internal IP of Freeswitch beghind the NAT.
> 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]*
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>
> 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
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> 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.*
>
> *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 gascagonzalo at gmail.com  Tue Feb 23 03:16:22 2016
From: gascagonzalo at gmail.com (Gonzalo Gasca Meza)
Date: Mon, 22 Feb 2016 16:16:22 -0800
Subject: [Freeswitch-users] Freeswitch libvpx2-dev compile error Debian 8
In-Reply-To: <CALXCt0qKs89MrSvGk7ckx9M+geTdX+BRzW5AniMVh+uJ4ht6kg@mail.gmail.com>
References: <CAL+DTgLM8q8qhpgVOzq_asfiGMn7Ub=1CQ1Hf3ztnPiyBPee3g@mail.gmail.com>
	<CALXCt0qKs89MrSvGk7ckx9M+geTdX+BRzW5AniMVh+uJ4ht6kg@mail.gmail.com>
Message-ID: <CAL+DTg+UpVaiCL5PFiMYp9zM5j-xUoL64EG5jjfWKGhR5g7Q7A@mail.gmail.com>

Just as a note
Recent builds are failing with similar error:

https://freeswitch.org/bamboo/browse/FS-TMB/latest


On Mon, Feb 22, 2016 at 3:12 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> As told in irc, you must install *unstable* repository for compiling
> latest git master.
>
> Check on www.freeswitch.conf/confluence the page for debian 8 jessie with
> updated instruction
> Il 22/Feb/2016 10:59, "Gonzalo Gasca Meza" <gascagonzalo at gmail.com> ha
> scritto:
>
>> Compiling latest FS version getting the following error when running make:
>>
>> You must install libvpx2-dev to build mod_av.  Stop.
>>
>> *Using this guide:*
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video
>>
>> *Version: **Debian 8.3 x64*
>>
>> *Troubleshooting*
>>
>> Previous version (Last week) was working fine in 2 different systems.
>>
>> *git log*
>>
>> commit 3bd26eaa6b2dff1d80cac4e58fc860fb96f92935
>>
>> Author: Anthony Minessale <anthm at freeswitch.org>
>>
>> Date:   Fri Feb 19 09:34:38 2016 -0600
>>
>>     FS-8847 #resolve [Silence Error on shutdown of video call]
>>
>> commit 9f4f67df5d2d9606ba88023b1bd05d7b88c69dad
>>
>> Author: Anthony Minessale <anthm at freeswitch.org>
>>
>> Date:   Thu Feb 18 17:04:01 2016 -0600
>>
>> *git diff*
>>
>>
>> https://freeswitch.org/stash/projects/FS/repos/freeswitch/diff/src/mod/applications/mod_av/Makefile.am?until=4ad0aa91a4a34fd10782dfe21aba74d49724f995
>>
>>
>> *logs (make)*
>>
>>
>> making all mod_amr
>>
>> make[4]: Entering directory
>> '/usr/src/freeswitch.git/src/mod/codecs/mod_amr'
>>
>> make  all-am
>>
>> make[5]: Entering directory
>> '/usr/src/freeswitch.git/src/mod/codecs/mod_amr'
>>
>>   CC       mod_amr_la-mod_amr.lo
>>
>>   CCLD     mod_amr.la
>>
>> make[5]: Leaving directory
>> '/usr/src/freeswitch.git/src/mod/codecs/mod_amr'
>>
>> make[4]: Leaving directory
>> '/usr/src/freeswitch.git/src/mod/codecs/mod_amr'
>>
>> making all mod_av
>>
>> make[4]: Entering directory
>> '/usr/src/freeswitch.git/src/mod/applications/mod_av'
>>
>> Makefile:915: *** You must install libvpx2-dev to build mod_av.  Stop.
>>
>> make[4]: Leaving directory
>> '/usr/src/freeswitch.git/src/mod/applications/mod_av'
>>
>> Makefile:648: recipe for target 'mod_av-all' failed
>>
>>
>> /usr/src/freeswitch.git/src/mod/applications/mod_av# tree
>>
>> .
>>
>> ??? avcodec.c
>>
>> ??? avformat.c
>>
>> ??? Makefile
>>
>> ??? Makefile.am
>>
>> ??? Makefile.in
>>
>> ??? mod_av.c
>>
>>
>> 0 directories, 6 files
>>
>> root at nkn-freeswitch-sfo1:/usr/src/freeswitch.git/src/mod/applications/mod_av#
>> ls -alh
>>
>> total 208K
>>
>> drwxr-xr-x  3 root root 4.0K Feb 22 04:46 .
>>
>> drwxr-xr-x 59 root root 4.0K Feb 22 04:20 ..
>>
>> -rw-r--r--  1 root root  46K Feb 22 04:20 avcodec.c
>>
>> -rw-r--r--  1 root root  63K Feb 22 04:20 avformat.c
>>
>> drwxr-xr-x  2 root root 4.0K Feb 22 04:24 .deps
>>
>> -rw-r--r--  1 root root  35K Feb 22 04:24 Makefile
>>
>> -rw-r--r--  1 root root  828 Feb 22 04:20 Makefile.am
>>
>> -rw-r--r--  1 root root  33K Feb 22 04:23 Makefile.in
>>
>> -rw-r--r--  1 root root 4.3K Feb 22 04:20 mod_av.c
>>
>> *Found*
>>
>>
>> http://lists.freeswitch.org/pipermail/freeswitch-users/2015-October/116401.html
>> But not sure how to proceed for this release
>>
>> 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 krice at freeswitch.org  Tue Feb 23 03:51:36 2016
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 22 Feb 2016 18:51:36 -0600
Subject: [Freeswitch-users] Freeswitch libvpx2-dev compile error Debian 8
In-Reply-To: <CAL+DTg+UpVaiCL5PFiMYp9zM5j-xUoL64EG5jjfWKGhR5g7Q7A@mail.gmail.com>
References: <CAL+DTgLM8q8qhpgVOzq_asfiGMn7Ub=1CQ1Hf3ztnPiyBPee3g@mail.gmail.com>
	<CALXCt0qKs89MrSvGk7ckx9M+geTdX+BRzW5AniMVh+uJ4ht6kg@mail.gmail.com>
	<CAL+DTg+UpVaiCL5PFiMYp9zM5j-xUoL64EG5jjfWKGhR5g7Q7A@mail.gmail.com>
Message-ID: <3CAF63BC-F20D-4CA8-A5B0-3E091BC7157A@freeswitch.org>

This is a known issue as this deb 7 build node is about to be replaced and will not be fixed. It just needs an upgraded libvpx2 there will be failures off and on like this in he short term as chrome is in the process of upgrading this lib and unfortunately its forcing us to do the same. This is the master branch so it can be broken at times.

Sent from my iPhone

> On Feb 22, 2016, at 6:16 PM, Gonzalo Gasca Meza <gascagonzalo at gmail.com> wrote:
> 
> Just as a note
> Recent builds are failing with similar error:
> 
> https://freeswitch.org/bamboo/browse/FS-TMB/latest
> 
> 
>> On Mon, Feb 22, 2016 at 3:12 AM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
>> As told in irc, you must install *unstable* repository for compiling latest git master.
>> 
>> Check on www.freeswitch.conf/confluence the page for debian 8 jessie with updated instruction
>> 
>> Il 22/Feb/2016 10:59, "Gonzalo Gasca Meza" <gascagonzalo at gmail.com> ha scritto:
>>> Compiling latest FS version getting the following error when running make:
>>> 
>>> You must install libvpx2-dev to build mod_av.  Stop.
>>> 
>>> Using this guide:
>>> 
>>> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video
>>> 
>>> Version: Debian 8.3 x64
>>> 
>>> Troubleshooting
>>> 
>>> Previous version (Last week) was working fine in 2 different systems.
>>> 
>>> git log
>>> 
>>> commit 3bd26eaa6b2dff1d80cac4e58fc860fb96f92935
>>> 
>>> Author: Anthony Minessale <anthm at freeswitch.org>
>>> 
>>> Date:   Fri Feb 19 09:34:38 2016 -0600
>>> 
>>>     FS-8847 #resolve [Silence Error on shutdown of video call]
>>> 
>>> commit 9f4f67df5d2d9606ba88023b1bd05d7b88c69dad
>>> 
>>> Author: Anthony Minessale <anthm at freeswitch.org>
>>> 
>>> 
>>> Date:   Thu Feb 18 17:04:01 2016 -0600
>>> 
>>> git diff
>>> 
>>> https://freeswitch.org/stash/projects/FS/repos/freeswitch/diff/src/mod/applications/mod_av/Makefile.am?until=4ad0aa91a4a34fd10782dfe21aba74d49724f995
>>> 
>>> 
>>> 
>>> logs (make)
>>> 
>>> 
>>> 
>>> making all mod_amr
>>> 
>>> make[4]: Entering directory '/usr/src/freeswitch.git/src/mod/codecs/mod_amr'
>>> 
>>> make  all-am
>>> 
>>> make[5]: Entering directory '/usr/src/freeswitch.git/src/mod/codecs/mod_amr'
>>> 
>>>   CC       mod_amr_la-mod_amr.lo
>>> 
>>>   CCLD     mod_amr.la
>>> 
>>> make[5]: Leaving directory '/usr/src/freeswitch.git/src/mod/codecs/mod_amr'
>>> 
>>> make[4]: Leaving directory '/usr/src/freeswitch.git/src/mod/codecs/mod_amr'
>>> 
>>> making all mod_av
>>> 
>>> make[4]: Entering directory '/usr/src/freeswitch.git/src/mod/applications/mod_av'
>>> 
>>> Makefile:915: *** You must install libvpx2-dev to build mod_av.  Stop.
>>> 
>>> make[4]: Leaving directory '/usr/src/freeswitch.git/src/mod/applications/mod_av'
>>> 
>>> Makefile:648: recipe for target 'mod_av-all' failed
>>> 
>>> 
>>> 
>>> /usr/src/freeswitch.git/src/mod/applications/mod_av# tree
>>> 
>>> .
>>> 
>>> ??? avcodec.c
>>> 
>>> ??? avformat.c
>>> 
>>> ??? Makefile
>>> 
>>> ??? Makefile.am
>>> 
>>> ??? Makefile.in
>>> 
>>> ??? mod_av.c
>>> 
>>> 
>>> 
>>> 0 directories, 6 files
>>> 
>>> root at nkn-freeswitch-sfo1:/usr/src/freeswitch.git/src/mod/applications/mod_av# ls -alh
>>> 
>>> total 208K
>>> 
>>> drwxr-xr-x  3 root root 4.0K Feb 22 04:46 .
>>> 
>>> drwxr-xr-x 59 root root 4.0K Feb 22 04:20 ..
>>> 
>>> -rw-r--r--  1 root root  46K Feb 22 04:20 avcodec.c
>>> 
>>> -rw-r--r--  1 root root  63K Feb 22 04:20 avformat.c
>>> 
>>> drwxr-xr-x  2 root root 4.0K Feb 22 04:24 .deps
>>> 
>>> -rw-r--r--  1 root root  35K Feb 22 04:24 Makefile
>>> 
>>> -rw-r--r--  1 root root  828 Feb 22 04:20 Makefile.am
>>> 
>>> -rw-r--r--  1 root root  33K Feb 22 04:23 Makefile.in
>>> 
>>> 
>>> -rw-r--r--  1 root root 4.3K Feb 22 04:20 mod_av.c
>>> 
>>> Found
>>> 
>>> 
>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2015-October/116401.html
>>> 
>>> But not sure how to proceed for this release
>>> 
>>> 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 s.safarov at gmail.com  Tue Feb 23 06:15:14 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 23 Feb 2016 03:15:14 +0000
Subject: [Freeswitch-users] Freeswitch libvpx2-dev compile error Debian 8
In-Reply-To: <3CAF63BC-F20D-4CA8-A5B0-3E091BC7157A@freeswitch.org>
References: <CAL+DTgLM8q8qhpgVOzq_asfiGMn7Ub=1CQ1Hf3ztnPiyBPee3g@mail.gmail.com>
	<CALXCt0qKs89MrSvGk7ckx9M+geTdX+BRzW5AniMVh+uJ4ht6kg@mail.gmail.com>
	<CAL+DTg+UpVaiCL5PFiMYp9zM5j-xUoL64EG5jjfWKGhR5g7Q7A@mail.gmail.com>
	<3CAF63BC-F20D-4CA8-A5B0-3E091BC7157A@freeswitch.org>
Message-ID: <CAHtxdDfEu2NWQpeaUWSSuUZ4XMKe9Z1gHVT3uDDFMOuOGf0wpA@mail.gmail.com>

Please check version of instaled libvpx2. Must be 1.5 according SD repo.

On Tue, Feb 23, 2016, 03:52 Ken Rice <krice at freeswitch.org> wrote:

> This is a known issue as this deb 7 build node is about to be replaced and
> will not be fixed. It just needs an upgraded libvpx2 there will be failures
> off and on like this in he short term as chrome is in the process of
> upgrading this lib and unfortunately its forcing us to do the same. This is
> the master branch so it can be broken at times.
>
> Sent from my iPhone
>
> On Feb 22, 2016, at 6:16 PM, Gonzalo Gasca Meza <gascagonzalo at gmail.com>
> wrote:
>
> Just as a note
> Recent builds are failing with similar error:
>
> https://freeswitch.org/bamboo/browse/FS-TMB/latest
>
>
> On Mon, Feb 22, 2016 at 3:12 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> As told in irc, you must install *unstable* repository for compiling
>> latest git master.
>>
>> Check on www.freeswitch.conf/confluence the page for debian 8 jessie
>> with updated instruction
>> Il 22/Feb/2016 10:59, "Gonzalo Gasca Meza" <gascagonzalo at gmail.com> ha
>> scritto:
>>
>>> Compiling latest FS version getting the following error when running
>>> make:
>>>
>>> You must install libvpx2-dev to build mod_av.  Stop.
>>>
>>> *Using this guide:*
>>>
>>> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video
>>>
>>> *Version: **Debian 8.3 x64*
>>>
>>> *Troubleshooting*
>>>
>>> Previous version (Last week) was working fine in 2 different systems.
>>>
>>> *git log*
>>>
>>> commit 3bd26eaa6b2dff1d80cac4e58fc860fb96f92935
>>>
>>> Author: Anthony Minessale <anthm at freeswitch.org>
>>>
>>> Date:   Fri Feb 19 09:34:38 2016 -0600
>>>
>>>     FS-8847 #resolve [Silence Error on shutdown of video call]
>>>
>>> commit 9f4f67df5d2d9606ba88023b1bd05d7b88c69dad
>>>
>>> Author: Anthony Minessale <anthm at freeswitch.org>
>>>
>>> Date:   Thu Feb 18 17:04:01 2016 -0600
>>>
>>> *git diff*
>>>
>>>
>>> https://freeswitch.org/stash/projects/FS/repos/freeswitch/diff/src/mod/applications/mod_av/Makefile.am?until=4ad0aa91a4a34fd10782dfe21aba74d49724f995
>>>
>>>
>>> *logs (make)*
>>>
>>>
>>> making all mod_amr
>>>
>>> make[4]: Entering directory
>>> '/usr/src/freeswitch.git/src/mod/codecs/mod_amr'
>>>
>>> make  all-am
>>>
>>> make[5]: Entering directory
>>> '/usr/src/freeswitch.git/src/mod/codecs/mod_amr'
>>>
>>>   CC       mod_amr_la-mod_amr.lo
>>>
>>>   CCLD     mod_amr.la
>>>
>>> make[5]: Leaving directory
>>> '/usr/src/freeswitch.git/src/mod/codecs/mod_amr'
>>>
>>> make[4]: Leaving directory
>>> '/usr/src/freeswitch.git/src/mod/codecs/mod_amr'
>>>
>>> making all mod_av
>>>
>>> make[4]: Entering directory
>>> '/usr/src/freeswitch.git/src/mod/applications/mod_av'
>>>
>>> Makefile:915: *** You must install libvpx2-dev to build mod_av.  Stop.
>>>
>>> make[4]: Leaving directory
>>> '/usr/src/freeswitch.git/src/mod/applications/mod_av'
>>>
>>> Makefile:648: recipe for target 'mod_av-all' failed
>>>
>>>
>>> /usr/src/freeswitch.git/src/mod/applications/mod_av# tree
>>>
>>> .
>>>
>>> ??? avcodec.c
>>>
>>> ??? avformat.c
>>>
>>> ??? Makefile
>>>
>>> ??? Makefile.am
>>>
>>> ??? Makefile.in
>>>
>>> ??? mod_av.c
>>>
>>>
>>> 0 directories, 6 files
>>>
>>> root at nkn-freeswitch-sfo1:/usr/src/freeswitch.git/src/mod/applications/mod_av#
>>> ls -alh
>>>
>>> total 208K
>>>
>>> drwxr-xr-x  3 root root 4.0K Feb 22 04:46 .
>>>
>>> drwxr-xr-x 59 root root 4.0K Feb 22 04:20 ..
>>>
>>> -rw-r--r--  1 root root  46K Feb 22 04:20 avcodec.c
>>>
>>> -rw-r--r--  1 root root  63K Feb 22 04:20 avformat.c
>>>
>>> drwxr-xr-x  2 root root 4.0K Feb 22 04:24 .deps
>>>
>>> -rw-r--r--  1 root root  35K Feb 22 04:24 Makefile
>>>
>>> -rw-r--r--  1 root root  828 Feb 22 04:20 Makefile.am
>>>
>>> -rw-r--r--  1 root root  33K Feb 22 04:23 Makefile.in
>>>
>>> -rw-r--r--  1 root root 4.3K Feb 22 04:20 mod_av.c
>>>
>>> *Found*
>>>
>>>
>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2015-October/116401.html
>>> But not sure how to proceed for this release
>>>
>>> Thanks
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From nandy1925 at gmail.com  Tue Feb 23 06:23:16 2016
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Tue, 23 Feb 2016 11:23:16 +0800
Subject: [Freeswitch-users] gsmopen problem: INCOMPATIBLE_DESTINATION
Message-ID: <CAHKqMb7MosEn_kA5KYhME+rfG5G+3Yqwiuoexanww=CFZb+N8g@mail.gmail.com>

Hello guys,

I installed mod_gsmopen. When testing incoming call, it hangs up with this
error message INCOMPATIBLE_DESTINATION.  I understand this is a codec
negotiation issue.

Looking at the CDR - the gsmopen presented L16 8000 in the SDP.  By
default, FS does early negotiation.

Snippet of my internal sip_profile:
<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="presence-privacy" value="$${presence_privacy}"/>
<param name="inbound-codec-negotiation" value="generous"/>

Take note: there is no disable-transcoding parameter which I understand FS
is transcodind.

In vars.xml:

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h
,G722,PCMU,PCMA,GSM"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>

I tried to add L16 at the end of global_codec_prefs which is unnecessary,
isn't it? But it didn't work, too.

Finally, I added L16 in the softphone, it worked.

I have installed gsmopen before and it worked (transcoding) with IP phones.

Is there anything I may have missed?  I appreciate for any leads.

Thank you,

/Nandy
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From rutu.patel at inextrix.com  Tue Feb 23 08:51:55 2016
From: rutu.patel at inextrix.com (Rutu Patel)
Date: Tue, 23 Feb 2016 11:21:55 +0530
Subject: [Freeswitch-users] Call dropping after 32 seconds
In-Reply-To: <0000015307b150a5-132a41d6-5bea-4af1-9686-2653df68a3b6-000000@email.amazonses.com>
References: <CAPQet1zs08bGNnGDVmsLH3R3VbvMCJW=8Cx3U+KPxwGpYSuzAA@mail.gmail.com>
	<0000015307b150a5-132a41d6-5bea-4af1-9686-2653df68a3b6-000000@email.amazonses.com>
Message-ID: <CAPQet1wJPNZj7S4ezzd=OSDqHzLKQMieOWvtL8Qm5ht1PcHyBw@mail.gmail.com>

Thanks for the reply.

Got your point about NATing issue and no response of 200 OK and as a
resoult ACK Timeout.
So, now to resolve the issue, if you can assist, what could be the possible
fixies?
>From where can i start? where to look?

Thanks.

--
Thanks,
Rutu Patel

<http://www.inextrix.com>

On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus <avi at avimarcus.net> wrote:

> 5 second response: 32 seconds is a timer/[network/NAT] issue.
>
> You have lots of 200s to the user since it's waiting for an ACK and keeps
> retrying, but for whatever network reason (router... sip alg?), it isn't
> getting one, so it triggers a timer to stop the call.
>
>
> -Avi Marcus
> BestFone
>
> On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel <rutu.patel at inextrix.com>
> wrote:
>
>> Hello,
>>
>> Having issue of call dropping after 32 seconds, here are the details-
>>
>> x.x.x.174: opensips server
>> x.x.x.166: freeswitch server
>> x.x.x.3:     another opensips server which is registered as gateway on
>> above freeswitch server
>> x.x.x.6: freeswitch server
>> x.x.x.47:  server through which the user is registered
>> I am trying to call from xxxx9 to xxxxxxx29858
>> xxxxxxx00181 is caller-id name and caller-id number
>>
>> Call flow is like this:
>> registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips
>> server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server)
>>
>> Outbound-proxy is set to x.x.x.174 in Gateway configuration.
>>
>> 1) Call is initiated by the user(xxxx9) registered with host x.x.x.174
>> 2) Call hit the freeswitch server x.x.x.166
>> 3) After '180 Ringing' and '183 Session Progress' packet
>> sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the
>> gateway x.x.x.3
>> But after 32 seconds call is dropped,
>> Within 32 seconds audio is ok from both end so it should not be the RTP
>> issue.
>> Here I have attached the file with sip logs, you can observer from the
>> file that, there are many '200 OK' from x.x.x.174 to x.x.x.166
>> and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then
>> 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped.
>>
>> What is wrong here? Any help would be appreciated here.
>>
>> Here is the file with sip logs
>> --
>> Thanks,
>> Rutu Patel
>> <http://www.inextrix.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 miha at softnet.si  Tue Feb 23 10:11:29 2016
From: miha at softnet.si (Miha)
Date: Tue, 23 Feb 2016 08:11:29 +0100
Subject: [Freeswitch-users] Freeswitch Centos 7 installation
Message-ID: <56CC0621.70607@softnet.si>

Hi,

i am having some problem with FS installation on Centos 7.

Here a log from "make":
making all mod_v8
make[4]: Entering directory `/usr/src/freeswitch/src/mod/languages/mod_v8'
cd /usr/src/freeswitch/libs/v8-3.24.14 && (test -f .stamp-patch || patch 
-t -p0 < /usr/src/freeswitch/src/mod/languages/mod_v8/v8-build. patch)
/bin/sh: patch: command not found
make[4]: *** [/usr/src/freeswitch/libs/v8-3.24.14/.stamp-patch] Error 127
make[4]: Leaving directory `/usr/src/freeswitch/src/mod/languages/mod_v8'
make[3]: *** [mod_v8-all] Error 1
make[3]: Leaving directory `/usr/src/freeswitch/src/mod'
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory `/usr/src/freeswitch/src'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/usr/src/freeswitch'
make: *** [all] Error 2

Could some point me on right direction to get is working?
tnx
miha
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From krice at freeswitch.org  Tue Feb 23 10:19:08 2016
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 23 Feb 2016 01:19:08 -0600
Subject: [Freeswitch-users] Freeswitch Centos 7 installation
In-Reply-To: <56CC0621.70607@softnet.si>
References: <56CC0621.70607@softnet.si>
Message-ID: <298201d16e0a$7cb24040$7616c0c0$@freeswitch.org>

The line ?/bin/sh: patch: command not found? is a big hint you might need to make sure you have patch installed

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Miha
Sent: Tuesday, February 23, 2016 1:11 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Freeswitch Centos 7 installation

 

Hi,

i am having some problem with FS installation on Centos 7.

Here a log from "make":
making all mod_v8
make[4]: Entering directory `/usr/src/freeswitch/src/mod/languages/mod_v8'
cd /usr/src/freeswitch/libs/v8-3.24.14 && (test -f .stamp-patch || patch -t -p0 < /usr/src/freeswitch/src/mod/languages/mod_v8/v8-build.                    patch)
/bin/sh: patch: command not found
make[4]: *** [/usr/src/freeswitch/libs/v8-3.24.14/.stamp-patch] Error 127
make[4]: Leaving directory `/usr/src/freeswitch/src/mod/languages/mod_v8'
make[3]: *** [mod_v8-all] Error 1
make[3]: Leaving directory `/usr/src/freeswitch/src/mod'
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory `/usr/src/freeswitch/src'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/usr/src/freeswitch'
make: *** [all] Error 2

Could some point me on right direction to get is working?
tnx
miha

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From jurij.ivo at gmail.com  Tue Feb 23 10:37:39 2016
From: jurij.ivo at gmail.com (Jurijs Ivolga)
Date: Tue, 23 Feb 2016 09:37:39 +0200
Subject: [Freeswitch-users] Call dropping after 32 seconds
In-Reply-To: <CAPQet1wJPNZj7S4ezzd=OSDqHzLKQMieOWvtL8Qm5ht1PcHyBw@mail.gmail.com>
References: <CAPQet1zs08bGNnGDVmsLH3R3VbvMCJW=8Cx3U+KPxwGpYSuzAA@mail.gmail.com>
	<0000015307b150a5-132a41d6-5bea-4af1-9686-2653df68a3b6-000000@email.amazonses.com>
	<CAPQet1wJPNZj7S4ezzd=OSDqHzLKQMieOWvtL8Qm5ht1PcHyBw@mail.gmail.com>
Message-ID: <CAGYpU64NgPRKPCywdNn3whBh8VH6=eYLy=PJPGrYn8JJ5iMPGQ@mail.gmail.com>

Hi,

1) You have very complex set-up and I doubt that you need it.

2) As far as you have user with ip x.x.x.174 and opensips server with same
ip x.x.x.174 it very hard to debug. So I propose you to send new log where
will be difference between user ip and opensips IP.

3) If you have possibility, try to register directly with a user to x.x.x.3
gateway and check if same issue still exists, if there is no such issue
anymore, then thee is definitely issue in your opensips x.x.x.174 and
freeswitch x.x.x.166. My point here is that you need to isolate issue and
to understand what part of your set-up works as expected and what is faulty.

With kind regards,

Jurijs

2016-02-23 7:51 GMT+02:00 Rutu Patel <rutu.patel at inextrix.com>:

> Thanks for the reply.
>
> Got your point about NATing issue and no response of 200 OK and as a
> resoult ACK Timeout.
> So, now to resolve the issue, if you can assist, what could be the
> possible fixies?
> From where can i start? where to look?
>
> Thanks.
>
> --
> Thanks,
> Rutu Patel
>
> <http://www.inextrix.com>
>
> On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> 5 second response: 32 seconds is a timer/[network/NAT] issue.
>>
>> You have lots of 200s to the user since it's waiting for an ACK and keeps
>> retrying, but for whatever network reason (router... sip alg?), it isn't
>> getting one, so it triggers a timer to stop the call.
>>
>>
>> -Avi Marcus
>> BestFone
>>
>> On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel <rutu.patel at inextrix.com>
>> wrote:
>>
>>> Hello,
>>>
>>> Having issue of call dropping after 32 seconds, here are the details-
>>>
>>> x.x.x.174: opensips server
>>> x.x.x.166: freeswitch server
>>> x.x.x.3:     another opensips server which is registered as gateway on
>>> above freeswitch server
>>> x.x.x.6: freeswitch server
>>> x.x.x.47:  server through which the user is registered
>>> I am trying to call from xxxx9 to xxxxxxx29858
>>> xxxxxxx00181 is caller-id name and caller-id number
>>>
>>> Call flow is like this:
>>> registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips
>>> server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server)
>>>
>>> Outbound-proxy is set to x.x.x.174 in Gateway configuration.
>>>
>>> 1) Call is initiated by the user(xxxx9) registered with host x.x.x.174
>>> 2) Call hit the freeswitch server x.x.x.166
>>> 3) After '180 Ringing' and '183 Session Progress' packet
>>> sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the
>>> gateway x.x.x.3
>>> But after 32 seconds call is dropped,
>>> Within 32 seconds audio is ok from both end so it should not be the RTP
>>> issue.
>>> Here I have attached the file with sip logs, you can observer from the
>>> file that, there are many '200 OK' from x.x.x.174 to x.x.x.166
>>> and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then
>>> 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped.
>>>
>>> What is wrong here? Any help would be appreciated here.
>>>
>>> Here is the file with sip logs
>>> --
>>> Thanks,
>>> Rutu Patel
>>> <http://www.inextrix.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


Jurijs

2016-02-23 7:51 GMT+02:00 Rutu Patel <rutu.patel at inextrix.com>:

> Thanks for the reply.
>
> Got your point about NATing issue and no response of 200 OK and as a
> resoult ACK Timeout.
> So, now to resolve the issue, if you can assist, what could be the
> possible fixies?
> From where can i start? where to look?
>
> Thanks.
>
> --
> Thanks,
> Rutu Patel
>
> <http://www.inextrix.com>
>
> On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> 5 second response: 32 seconds is a timer/[network/NAT] issue.
>>
>> You have lots of 200s to the user since it's waiting for an ACK and keeps
>> retrying, but for whatever network reason (router... sip alg?), it isn't
>> getting one, so it triggers a timer to stop the call.
>>
>>
>> -Avi Marcus
>> BestFone
>>
>> On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel <rutu.patel at inextrix.com>
>> wrote:
>>
>>> Hello,
>>>
>>> Having issue of call dropping after 32 seconds, here are the details-
>>>
>>> x.x.x.174: opensips server
>>> x.x.x.166: freeswitch server
>>> x.x.x.3:     another opensips server which is registered as gateway on
>>> above freeswitch server
>>> x.x.x.6: freeswitch server
>>> x.x.x.47:  server through which the user is registered
>>> I am trying to call from xxxx9 to xxxxxxx29858
>>> xxxxxxx00181 is caller-id name and caller-id number
>>>
>>> Call flow is like this:
>>> registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips
>>> server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server)
>>>
>>> Outbound-proxy is set to x.x.x.174 in Gateway configuration.
>>>
>>> 1) Call is initiated by the user(xxxx9) registered with host x.x.x.174
>>> 2) Call hit the freeswitch server x.x.x.166
>>> 3) After '180 Ringing' and '183 Session Progress' packet
>>> sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the
>>> gateway x.x.x.3
>>> But after 32 seconds call is dropped,
>>> Within 32 seconds audio is ok from both end so it should not be the RTP
>>> issue.
>>> Here I have attached the file with sip logs, you can observer from the
>>> file that, there are many '200 OK' from x.x.x.174 to x.x.x.166
>>> and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then
>>> 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped.
>>>
>>> What is wrong here? Any help would be appreciated here.
>>>
>>> Here is the file with sip logs
>>> --
>>> Thanks,
>>> Rutu Patel
>>> <http://www.inextrix.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb 23 11:40:16 2016
From: lte at lte-net.de (Fred Schulz)
Date: Tue, 23 Feb 2016 09:40:16 +0100
Subject: [Freeswitch-users] mod_com_g729 validator
Message-ID: <010F1A96-67F0-432A-AB90-79383BC86FB7@lte-net.de>

Hi all,

does one know if it is possible to pass parameters towards the validator?
We plan to install not just only one system, so we?re trying to use a script. Therefore it would be easier to install our systems.

Thanks
Fred

From miha at softnet.si  Tue Feb 23 11:49:34 2016
From: miha at softnet.si (Miha)
Date: Tue, 23 Feb 2016 09:49:34 +0100
Subject: [Freeswitch-users] Freeswitch Centos 7 installation
In-Reply-To: <298201d16e0a$7cb24040$7616c0c0$@freeswitch.org>
References: <56CC0621.70607@softnet.si>
	<298201d16e0a$7cb24040$7616c0c0$@freeswitch.org>
Message-ID: <56CC1D1E.6040504@softnet.si>

Tnx Ken:)

I should i known that :)


br
miha

On 23/02/2016 08:19, Ken Rice wrote:
>
> The line ?/bin/sh: patch: command not found? is a big hint you might 
> need to make sure you have patch installed
>
> *From:*freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Miha
> *Sent:* Tuesday, February 23, 2016 1:11 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] Freeswitch Centos 7 installation
>
> Hi,
>
> i am having some problem with FS installation on Centos 7.
>
> Here a log from "make":
> making all mod_v8
> make[4]: Entering directory `/usr/src/freeswitch/src/mod/languages/mod_v8'
> cd /usr/src/freeswitch/libs/v8-3.24.14 && (test -f .stamp-patch || 
> patch -t -p0 < /usr/src/freeswitch/src/mod/languages/mod_v8/v8-build. 
> patch)
> /bin/sh: patch: command not found
> make[4]: *** [/usr/src/freeswitch/libs/v8-3.24.14/.stamp-patch] Error 127
> make[4]: Leaving directory `/usr/src/freeswitch/src/mod/languages/mod_v8'
> make[3]: *** [mod_v8-all] Error 1
> make[3]: Leaving directory `/usr/src/freeswitch/src/mod'
> make[2]: *** [all-recursive] Error 1
> make[2]: Leaving directory `/usr/src/freeswitch/src'
> make[1]: *** [all-recursive] Error 1
> make[1]: Leaving directory `/usr/src/freeswitch'
> make: *** [all] Error 2
>
> Could some point me on right direction to get is working?
> tnx
> miha
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb 23 16:34:22 2016
From: aqsyounas at gmail.com (Aqs Younas)
Date: Tue, 23 Feb 2016 18:34:22 +0500
Subject: [Freeswitch-users] Unable to detect beep with mod_avmd
Message-ID: <CAGVOjVUvrPVbO_jh97f0gKcubbOonvDzpo-7=e3_0i3frSBZfA@mail.gmail.com>

Hi, I am using avmd to detect beep over a channel. Beep is generated by
below script.

<extension name="10014">
       <condition field="destination_number" expression="^1001$">
        <action application="playback"
data="tone_stream://L=5;%(500,6850,850)"/>
         <action application="hangup"/>
       </condition>
  </extension>

and starts avmd on fs_cli as.

freeswitch at debian>avmd 30d99731-4ff3-4f02-ad4b-85a3a8b9d600 start


I could proper hear beeps over the phone but does not see avmd detecting
this. Does not see anything like "beep detected" in logs.

I am made several attempts but unable to detect beep even a single time.

Is there something wrong that i am doing?

Thanks in advance.
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From aubalde at presenceco.com  Tue Feb 23 11:15:41 2016
From: aubalde at presenceco.com (=?UTF-8?Q?Agust=C3=AD_Ubalde?=)
Date: Tue, 23 Feb 2016 09:15:41 +0100
Subject: [Freeswitch-users] Invalid sdp on WebRTC connections
In-Reply-To: <CAKbxfG-0WAemipjMZHb96HvV9=NDa+-pVRVWMeteSy-G5RD1hw@mail.gmail.com>
References: <CAKThY06c7v=ftYa=qX93UHkUJUvTcPJq0B5pmH3SoDVfJFuWPw@mail.gmail.com>
	<CAKbxfG-0WAemipjMZHb96HvV9=NDa+-pVRVWMeteSy-G5RD1hw@mail.gmail.com>
Message-ID: <CAKThY04T7Hc7kL7dB3XxC_mvYiszgJV6WDCdL2aZmoAA2RLtsA@mail.gmail.com>

Hi,

*ext-rtp-ip* is our public ip address.
*apply-candidate-acl* is not defined at internal.xml.


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/>

2016-02-22 20:15 GMT+01:00 Anthony Minessale <anthony.minessale at gmail.com>:

> See
>
> ext-rtp-ip and
>
> apply-candidate-acl
>
>
>
> On Mon, Feb 22, 2016 at 9:51 AM, Agust? Ubalde <aubalde at presenceco.com>
> wrote:
>
>> We have a WebRTC environment with Freeswitch and sipml5. The fact is that
>> when a WebRTC extension receives a call, the SDP that sends Freeswitch not
>> show the correct candidate:
>> *======================================================================*
>> *22d33210-d709-11e5-bdcd-078a69e08f30 Local SDP:*
>> *22d33210-d709-11e5-bdcd-078a69e08f30 v=0*
>> *22d33210-d709-11e5-bdcd-078a69e08f30 o=FreeSWITCH 1455870972 1455870973
>> IN IP4 X.X.X.X*
>> *22d33210-d709-11e5-bdcd-078a69e08f30 s=FreeSWITCH*
>> *22d33210-d709-11e5-bdcd-078a69e08f30 c=IN IP4 X.X.X.X*
>> *======================================================================*
>>
>> Where X.X.X.X is the internal IP of Freeswitch beghind the NAT.
>> 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
>>
>
>
>
> --
> 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 brian at freeswitch.org  Tue Feb 23 17:44:01 2016
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Feb 2016 08:44:01 -0600
Subject: [Freeswitch-users] Invalid sdp on WebRTC connections
In-Reply-To: <CAKThY04T7Hc7kL7dB3XxC_mvYiszgJV6WDCdL2aZmoAA2RLtsA@mail.gmail.com>
References: <CAKThY06c7v=ftYa=qX93UHkUJUvTcPJq0B5pmH3SoDVfJFuWPw@mail.gmail.com>
	<CAKbxfG-0WAemipjMZHb96HvV9=NDa+-pVRVWMeteSy-G5RD1hw@mail.gmail.com>
	<CAKThY04T7Hc7kL7dB3XxC_mvYiszgJV6WDCdL2aZmoAA2RLtsA@mail.gmail.com>
Message-ID: <CAEJMVkC8gLqwYNBaMbbz0GgQ6bUT6-+u-uU5pZVZKdwm9efZ=g@mail.gmail.com>

I think he's stating that you need to define apply-candidate-acl

On Tue, Feb 23, 2016 at 2:15 AM, Agust? Ubalde <aubalde at presenceco.com>
wrote:

> Hi,
>
> *ext-rtp-ip* is our public ip address.
> *apply-candidate-acl* is not defined at internal.xml.
>
>
> 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/>
>
> 2016-02-22 20:15 GMT+01:00 Anthony Minessale <anthony.minessale at gmail.com>
> :
>
>> See
>>
>> ext-rtp-ip and
>>
>> apply-candidate-acl
>>
>>
>>
>> On Mon, Feb 22, 2016 at 9:51 AM, Agust? Ubalde <aubalde at presenceco.com>
>> wrote:
>>
>>> We have a WebRTC environment with Freeswitch and sipml5. The fact is
>>> that when a WebRTC extension receives a call, the SDP that sends Freeswitch
>>> not show the correct candidate:
>>> *======================================================================*
>>> *22d33210-d709-11e5-bdcd-078a69e08f30 Local SDP:*
>>> *22d33210-d709-11e5-bdcd-078a69e08f30 v=0*
>>> *22d33210-d709-11e5-bdcd-078a69e08f30 o=FreeSWITCH 1455870972 1455870973
>>> IN IP4 X.X.X.X*
>>> *22d33210-d709-11e5-bdcd-078a69e08f30 s=FreeSWITCH*
>>> *22d33210-d709-11e5-bdcd-078a69e08f30 c=IN IP4 X.X.X.X*
>>> *======================================================================*
>>>
>>> Where X.X.X.X is the internal IP of Freeswitch beghind the NAT.
>>> 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
>>>
>>
>>
>>
>> --
>> 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/>*
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

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 anthony.minessale at gmail.com  Tue Feb 23 19:18:24 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 23 Feb 2016 10:18:24 -0600
Subject: [Freeswitch-users] Invalid sdp on WebRTC connections
In-Reply-To: <CAEJMVkC8gLqwYNBaMbbz0GgQ6bUT6-+u-uU5pZVZKdwm9efZ=g@mail.gmail.com>
References: <CAKThY06c7v=ftYa=qX93UHkUJUvTcPJq0B5pmH3SoDVfJFuWPw@mail.gmail.com>
	<CAKbxfG-0WAemipjMZHb96HvV9=NDa+-pVRVWMeteSy-G5RD1hw@mail.gmail.com>
	<CAKThY04T7Hc7kL7dB3XxC_mvYiszgJV6WDCdL2aZmoAA2RLtsA@mail.gmail.com>
	<CAEJMVkC8gLqwYNBaMbbz0GgQ6bUT6-+u-uU5pZVZKdwm9efZ=g@mail.gmail.com>
Message-ID: <CAKbxfG9qa9oYjY3AFM+oLtCp+a8mEb-AnR7uvX-wQ3tZy_sJrA@mail.gmail.com>

apply-candidate-acl controls the networks the candidates will be preferred
from.
They are parsed in order they are defined.


On Tue, Feb 23, 2016 at 8:44 AM, Brian West <brian at freeswitch.org> wrote:

> I think he's stating that you need to define apply-candidate-acl
>
> On Tue, Feb 23, 2016 at 2:15 AM, Agust? Ubalde <aubalde at presenceco.com>
> wrote:
>
>> Hi,
>>
>> *ext-rtp-ip* is our public ip address.
>> *apply-candidate-acl* is not defined at internal.xml.
>>
>>
>> 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/>
>>
>> 2016-02-22 20:15 GMT+01:00 Anthony Minessale <anthony.minessale at gmail.com
>> >:
>>
>>> See
>>>
>>> ext-rtp-ip and
>>>
>>> apply-candidate-acl
>>>
>>>
>>>
>>> On Mon, Feb 22, 2016 at 9:51 AM, Agust? Ubalde <aubalde at presenceco.com>
>>> wrote:
>>>
>>>> We have a WebRTC environment with Freeswitch and sipml5. The fact is
>>>> that when a WebRTC extension receives a call, the SDP that sends Freeswitch
>>>> not show the correct candidate:
>>>> *======================================================================*
>>>> *22d33210-d709-11e5-bdcd-078a69e08f30 Local SDP:*
>>>> *22d33210-d709-11e5-bdcd-078a69e08f30 v=0*
>>>> *22d33210-d709-11e5-bdcd-078a69e08f30 o=FreeSWITCH 1455870972
>>>> 1455870973 IN IP4 X.X.X.X*
>>>> *22d33210-d709-11e5-bdcd-078a69e08f30 s=FreeSWITCH*
>>>> *22d33210-d709-11e5-bdcd-078a69e08f30 c=IN IP4 X.X.X.X*
>>>> *======================================================================*
>>>>
>>>> Where X.X.X.X is the internal IP of Freeswitch beghind the NAT.
>>>> 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
>>>>
>>>
>>>
>>>
>>> --
>>> 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/>*
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> 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
>



-- 
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 steveayre at gmail.com  Tue Feb 23 19:38:16 2016
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 23 Feb 2016 16:38:16 +0000
Subject: [Freeswitch-users] mod_com_g729 validator
In-Reply-To: <010F1A96-67F0-432A-AB90-79383BC86FB7@lte-net.de>
References: <010F1A96-67F0-432A-AB90-79383BC86FB7@lte-net.de>
Message-ID: <CAFiqYumDSz+jZ8QfgpU4y0FSO5hA2EsWSG9tyqe6r=reSJVONg@mail.gmail.com>

You might be able to interact with it using expect

On 23 February 2016 at 08:40, Fred Schulz <lte at lte-net.de> wrote:

> Hi all,
>
> does one know if it is possible to pass parameters towards the validator?
> We plan to install not just only one system, so we?re trying to use a
> script. Therefore it would be easier to install our systems.
>
> Thanks
> Fred
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Tue Feb 23 20:00:40 2016
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Feb 2016 11:00:40 -0600
Subject: [Freeswitch-users] mod_com_g729 validator
In-Reply-To: <010F1A96-67F0-432A-AB90-79383BC86FB7@lte-net.de>
References: <010F1A96-67F0-432A-AB90-79383BC86FB7@lte-net.de>
Message-ID: <CAEJMVkDe6ErKsjSjN5f59AMdCxePoMKArSrFE5rbx81o6hmmgw@mail.gmail.com>

https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/scripts/perl/g729_activate


On Tue, Feb 23, 2016 at 2:40 AM, Fred Schulz <lte at lte-net.de> wrote:

> Hi all,
>
> does one know if it is possible to pass parameters towards the validator?
> We plan to install not just only one system, so we?re trying to use a
> script. Therefore it would be easier to install our systems.
>
> Thanks
> Fred
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

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 lte at lte-net.de  Tue Feb 23 21:36:01 2016
From: lte at lte-net.de (Fred Schulz)
Date: Tue, 23 Feb 2016 19:36:01 +0100
Subject: [Freeswitch-users] mod_com_g729 validator
In-Reply-To: <CAEJMVkDe6ErKsjSjN5f59AMdCxePoMKArSrFE5rbx81o6hmmgw@mail.gmail.com>
References: <010F1A96-67F0-432A-AB90-79383BC86FB7@lte-net.de>
	<CAEJMVkDe6ErKsjSjN5f59AMdCxePoMKArSrFE5rbx81o6hmmgw@mail.gmail.com>
Message-ID: <8A481BE2-D52A-412C-9D75-F4C6E69EEEAB@lte-net.de>

Thanks

I give it a try :)

> Am 23.02.2016 um 18:00 schrieb Brian West <brian at freeswitch.org>:
> 
> https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/scripts/perl/g729_activate <https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/scripts/perl/g729_activate>
> 
> 
> On Tue, Feb 23, 2016 at 2:40 AM, Fred Schulz <lte at lte-net.de <mailto:lte at lte-net.de>> wrote:
> Hi all,
> 
> does one know if it is possible to pass parameters towards the validator?
> We plan to install not just only one system, so we?re trying to use a script. Therefore it would be easier to install our systems.
> 
> Thanks
> Fred
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> -- 
> Brian West
> brian at freeswitch.org <mailto:brian at freeswitch.org>
> 
> Twitter: @FreeSWITCH , @briankwest
> http://www.freeswitchbook.com <http://www.freeswitchbook.com/>
> http://www.freeswitchcookbook.com <http://www.freeswitchcookbook.com/>
> 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 victor.chukalovskiy at gmail.com  Tue Feb 23 21:52:28 2016
From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy)
Date: Tue, 23 Feb 2016 13:52:28 -0500
Subject: [Freeswitch-users] Incorrect reply to T.38 re-INVITE
In-Reply-To: <op.yc3hjevstbe8eu@ot.chello.pl>
References: <op.yc3hjevstbe8eu@ot.chello.pl>
Message-ID: <56CCAA6C.3010901@gmail.com>

Hi Tomasz,

Hi, I was the guy who reported and followed-up on 
https://freeswitch.org/jira/browse/FS-6954

What I recall that it needed to be fixed for a variety of possible 
config call flow scenarios, that can be represented as 3 independent 
factors:

Factor 1 - whether FS is in proxy_media, bypass_media, or default full 
media mode. (3 options)
Factor 2 - direction in which t.38 reINVITE flows (A --> B or B --> A). 
(2 options)
Factor 3 - whether A leg of the call and B leg of the call is the same 
SIP profile or two different SIP profiles. (2 options)

So, you have a matrix representing possible scenarios based on these 3 
independent factors...So, you have at least 3 x 2 x 2 = 12 distinct 
cases to test. Bugfixes were done for some of them, but there were some 
cases left where bug fix was never complete, at least not until I 
gave-up on that JIRA. I'd imagine cases that were fixed made it's way to 
1.6, while cases that were not fixed can still be in the same state.

If you want to be confident about current state, I'd recommend getting 
1.6 and setting test environment where you can verify each possible 
configuration + call flow scenario.

Cheers,
-Victor

On 16-02-19 02:24 PM, Tomasz Ostrowski wrote:
> Hello,
> while testing FreeSWITCH 1.4.26 I stumbled upon old and seemingly ignored
> problem when FreeSWITCH replies with SDP containing:
>
> m=image 0 udptl t38
> m=image 0 udptl t38
>
> when receiving re-INVITE with disabled audio media and enabled image media
> (switching from audio to image). As far as I know this is correct way to
> change media, from RFC 3264:
>
> 8.1 Adding a Media Stream
>
>      New media streams are created by new additional media descriptions
>      below the existing ones, or by reusing the "slot" used by an old
>      media stream which had been disabled by setting its port to zero.
>
>      Reusing its slot means that the new media description replaces the
>      old one, but retains its positioning relative to other media
>      descriptions in  the SDP.  New media descriptions MUST appear below
>      any existing media sections.  The rules for formatting these media
>      descriptions are identical to those described in Section 5.
>
>      When the answerer receives an SDP with more media descriptions than
>      the previous SDP from the offerer, or it receives an SDP with a media
>      stream in a slot where the port was previously zero, the answerer
>      knows that new media streams are being added.  These can be rejected
>      or accepted by placing an appropriately structured media description
>      in the answer.  The procedures for constructing the new media
>      description in the answer are described in Section 6.
>
> This problem is mentioned in:
> https://freeswitch.org/jira/browse/FS-7037
> https://freeswitch.org/jira/browse/FS-6212
>
> With reversed transmission direction (when FreeSWITCH receives FAX and
> re-INVITES) re-INVITE contains only image media, but this is easier to
> accept (https://freeswitch.org/jira/browse/FS-6954 -  correcting it caused
> interoperability problems between FS versions), while not accepting
> correct SDP by FreeSWITCH is really painful as it requires implementing
> RFC non-compliant negotiation and probably adding special switch in
> configuration to be interoperable.
>
> Is this issue fixed in FreeSWITCH 1.6 (I cannot find any further
> references in jira)?
> Could you give me any suggestions where to look in FreeSWITCH source code
> (10k LOC sofia.c seems pretty complex)?
>



From schoch+freeswitch.org at xwin32.com  Wed Feb 24 00:28:01 2016
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Tue, 23 Feb 2016 13:28:01 -0800
Subject: [Freeswitch-users] Multiple Choice Caller-ID
Message-ID: <CAOkuKddManZTowKD6F4Y_fm06KKH0oRD_tX_-UiFB9bLHXtRpg@mail.gmail.com>

We use the Polycom SoundPoint IP phones in the office, which work great.
One user has 3 different numbers to reach that phone (one is the main
office number with an extension, the other 2 ring direct).
These phones have 2 line buttons. I would like this user to be able to
choose from 3 different Caller-ID numbers when placing a call. I already do
this with 2 numbers by using the 2 different line buttons, but I'm trying
to figure out a way to choose a 3rd number when placing a call.
Any ideas will be appreciated.

-- 
Steve
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From brian at freeswitch.org  Wed Feb 24 00:49:33 2016
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Feb 2016 15:49:33 -0600
Subject: [Freeswitch-users] Multiple Choice Caller-ID
In-Reply-To: <CAOkuKddManZTowKD6F4Y_fm06KKH0oRD_tX_-UiFB9bLHXtRpg@mail.gmail.com>
References: <CAOkuKddManZTowKD6F4Y_fm06KKH0oRD_tX_-UiFB9bLHXtRpg@mail.gmail.com>
Message-ID: <CAEJMVkC-YHmxzYukq3su6oxTa8hvgaGTMwaUbb2w0cn3g18D7w@mail.gmail.com>

prefix dialing.

On Tue, Feb 23, 2016 at 3:28 PM, Steven Schoch <
schoch+freeswitch.org at xwin32.com> wrote:

> We use the Polycom SoundPoint IP phones in the office, which work great.
> One user has 3 different numbers to reach that phone (one is the main
> office number with an extension, the other 2 ring direct).
> These phones have 2 line buttons. I would like this user to be able to
> choose from 3 different Caller-ID numbers when placing a call. I already do
> this with 2 numbers by using the 2 different line buttons, but I'm trying
> to figure out a way to choose a 3rd number when placing a call.
> Any ideas will be appreciated.
>
> --
> 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
>



-- 

*Brian West*
brian at freeswitch.org


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

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 ssinyagin at gmail.com  Wed Feb 24 02:04:50 2016
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Wed, 24 Feb 2016 00:04:50 +0100
Subject: [Freeswitch-users] Multiple Choice Caller-ID
In-Reply-To: <CAEJMVkC-YHmxzYukq3su6oxTa8hvgaGTMwaUbb2w0cn3g18D7w@mail.gmail.com>
References: <CAOkuKddManZTowKD6F4Y_fm06KKH0oRD_tX_-UiFB9bLHXtRpg@mail.gmail.com>
	<CAEJMVkC-YHmxzYukq3su6oxTa8hvgaGTMwaUbb2w0cn3g18D7w@mail.gmail.com>
Message-ID: <CACOYK=iT3U7gaij2Z-V7qE=4H8xgnv4u9GEW23D5_wR+LgchXw@mail.gmail.com>

or an IVR that asks you which caller ID to use. Also it may remember your
previous choice or use some kind of a rule in your address book...

I've done this with a Perl script that looks up in Google Contacts, and the
contacts are organized into groups.


On Tue, Feb 23, 2016 at 10:49 PM, Brian West <brian at freeswitch.org> wrote:

> prefix dialing.
>
> On Tue, Feb 23, 2016 at 3:28 PM, Steven Schoch <
> schoch+freeswitch.org at xwin32.com> wrote:
>
>> We use the Polycom SoundPoint IP phones in the office, which work great.
>> One user has 3 different numbers to reach that phone (one is the main
>> office number with an extension, the other 2 ring direct).
>> These phones have 2 line buttons. I would like this user to be able to
>> choose from 3 different Caller-ID numbers when placing a call. I already do
>> this with 2 numbers by using the 2 different line buttons, but I'm trying
>> to figure out a way to choose a 3rd number when placing a call.
>> Any ideas will be appreciated.
>>
>> --
>> 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
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> 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 msc at freeswitch.org  Wed Feb 24 03:44:21 2016
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 23 Feb 2016 16:44:21 -0800
Subject: [Freeswitch-users] Multiple Choice Caller-ID
In-Reply-To: <CACOYK=iT3U7gaij2Z-V7qE=4H8xgnv4u9GEW23D5_wR+LgchXw@mail.gmail.com>
References: <CAOkuKddManZTowKD6F4Y_fm06KKH0oRD_tX_-UiFB9bLHXtRpg@mail.gmail.com>
	<CAEJMVkC-YHmxzYukq3su6oxTa8hvgaGTMwaUbb2w0cn3g18D7w@mail.gmail.com>
	<CACOYK=iT3U7gaij2Z-V7qE=4H8xgnv4u9GEW23D5_wR+LgchXw@mail.gmail.com>
Message-ID: <CAKzWOxU9PV=ecp_4=v1p=6Xp6O-yvzVS_9A-_KaYrmnruV7b_g@mail.gmail.com>

On Tue, Feb 23, 2016 at 3:04 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> or an IVR that asks you which caller ID to use. Also it may remember your
> previous choice or use some kind of a rule in your address book...
>
> I've done this with a Perl script that looks up in Google Contacts, and
> the contacts are organized into groups.
>
>
>
> On Tue, Feb 23, 2016 at 10:49 PM, Brian West <brian at freeswitch.org> wrote:
>
>> prefix dialing.
>>
> +1

Once the user gets accustomed to dialing 7+phone number for caller ID 123
and 8+phone number for caller ID 456 then they'll never go back. (Use
whatever prefixes are the best fit for your existing dialing plan.)

-MSC
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From msc at freeswitch.org  Wed Feb 24 04:27:12 2016
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 23 Feb 2016 17:27:12 -0800
Subject: [Freeswitch-users] Unable to detect beep with mod_avmd
In-Reply-To: <CAGVOjVUvrPVbO_jh97f0gKcubbOonvDzpo-7=e3_0i3frSBZfA@mail.gmail.com>
References: <CAGVOjVUvrPVbO_jh97f0gKcubbOonvDzpo-7=e3_0i3frSBZfA@mail.gmail.com>
Message-ID: <CAKzWOxXVV5fjyOEP=3mDbNH29Rv2=7iesPbdPZkzypriPVcWzg@mail.gmail.com>

It definitely works, I just don't think it works on a one-legged call. From
fs_cli try this:

originate loopback/1001 &playback(silence_stream://60000)

You'll have two uuids. Try the avmd app on each one. It will detect it on
the "outbound" leg but not the "inbound" leg.

-MSC

On Tue, Feb 23, 2016 at 5:34 AM, Aqs Younas <aqsyounas at gmail.com> wrote:

> Hi, I am using avmd to detect beep over a channel. Beep is generated by
> below script.
>
> <extension name="10014">
>        <condition field="destination_number" expression="^1001$">
>         <action application="playback"
> data="tone_stream://L=5;%(500,6850,850)"/>
>          <action application="hangup"/>
>        </condition>
>   </extension>
>
> and starts avmd on fs_cli as.
>
> freeswitch at debian>avmd 30d99731-4ff3-4f02-ad4b-85a3a8b9d600 start
>
>
> I could proper hear beeps over the phone but does not see avmd detecting
> this. Does not see anything like "beep detected" in logs.
>
> I am made several attempts but unable to detect beep even a single time.
>
> Is there something wrong that i am doing?
>
> Thanks in advance.
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Wed Feb 24 14:40:40 2016
From: naveen32india at gmail.com (Naveen Tamanam)
Date: Wed, 24 Feb 2016 17:10:40 +0530
Subject: [Freeswitch-users] Do we have option invalid-sound-long in
	freeswitch IVR menu ?
Message-ID: <CABsWbRbPLR=KFx0rrJtuU28Kr86bQfVHbsoqN0gaLwzsRVTXxw@mail.gmail.com>

I would like to play different invalid sound for the first failure which
would be long sound and some other invalid sound for next retries.

Do we have any kind of option like greet-long and greet-short to play
different invalid sounds, like invalid-long invalid-short ?

-- 
Thanks & Regards,
Naveen Tamanam
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From veerabhadrarao.kankatala at panamaxil.com  Wed Feb 24 14:55:00 2016
From: veerabhadrarao.kankatala at panamaxil.com (Veerabhadrarao Kankatala)
Date: Wed, 24 Feb 2016 06:55:00 -0500 (EST)
Subject: [Freeswitch-users] libyuv problem
Message-ID: <1700116939.11698256.1456314900112.JavaMail.zimbra@panamaxil.com>

hello, 

I am getting following error while configrung freeswitch 1.6.5 

./configure (success) 

but when i run make command i am facing following error 

making all mod_fsv 
make[4]: Entering directory `/usr/src/bhadra/freeswitch-1.6.5/src/mod/applications/mod_fsv' 
Makefile:797: *** You must install libyuv-dev to build mod_fsv. Stop. 
make[4]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5/src/mod/applications/mod_fsv' 
make[3]: *** [mod_fsv-all] Error 1 
make[3]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5/src/mod' 
make[2]: *** [all-recursive] Error 1 
make[2]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5/src' 
make[1]: *** [all-recursive] Error 1 
make[1]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5' 
make: *** [all] Error 2 

but i already installed libyuv packages 

yum install libyuv-devel 
Loaded plugins: langpacks, presto, refresh-packagekit 
Package libyuv-devel-0-0.17.20121221svn522.fc18.x86_64 already installed and latest version 
Nothing to do 


i would like to know which version of libyuv, libyuv-devel is require for freeswitch 1.6.5V 

I am using Fedora 18 64Bit, 

Please help me to come out from this problem 

Thanks in advance 

-- 
Thanks & Regards 
Veerabhadrarao Kankatala 
Software Developer (C & Unix ) 
PANAMAX INFOTECH LIMITED 
Mobile: +91-8401231249 
Messenger Id: Skype: veerabhadrarao.kankatala 
E-mail: veerabhadrarao.kankatala at panamaxil.com 

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From bhadrarao.kankatala at gmail.com  Wed Feb 24 14:47:49 2016
From: bhadrarao.kankatala at gmail.com (Kankatala Bhadra Rao)
Date: Wed, 24 Feb 2016 17:17:49 +0530
Subject: [Freeswitch-users] Lib yuv problem
Message-ID: <CAPndX91XuAPEf0N6RRsMkVjLpErvYSWydWOD44kVe8cX5Oa3-g@mail.gmail.com>

hello,

I am getting following error while configrung freeswitch 1.6.5

./configure (success)

but when i run make command i am facing following error

making all mod_fsv
make[4]: Entering directory
`/usr/src/bhadra/freeswitch-1.6.5/src/mod/applications/mod_fsv'
Makefile:797: *** You must install libyuv-dev to build mod_fsv. Stop.
make[4]: Leaving directory
`/usr/src/bhadra/freeswitch-1.6.5/src/mod/applications/mod_fsv'
make[3]: *** [mod_fsv-all] Error 1
make[3]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5/src/mod'
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5/src'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5'
make: *** [all] Error 2

but i already installed libyuv packages

yum install libyuv-devel
Loaded plugins: langpacks, presto, refresh-packagekit
Package libyuv-devel-0-0.17.20121221svn522.fc18.x86_64 already installed
and latest version
Nothing to do

i would like to know which version oflibyuv, libyuv-devel  is require for
freeswitch 1.6.5V

I am using Fedora 18 64Bit,

Please help me to come out from this problem

Thanks in advance
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From mike at jerris.com  Wed Feb 24 16:19:30 2016
From: mike at jerris.com (Michael Jerris)
Date: Wed, 24 Feb 2016 08:19:30 -0500
Subject: [Freeswitch-users] Do we have option invalid-sound-long in
 freeswitch IVR menu ?
In-Reply-To: <CABsWbRbPLR=KFx0rrJtuU28Kr86bQfVHbsoqN0gaLwzsRVTXxw@mail.gmail.com>
References: <CABsWbRbPLR=KFx0rrJtuU28Kr86bQfVHbsoqN0gaLwzsRVTXxw@mail.gmail.com>
Message-ID: <CAEwTsAhe8NDi-+qizFC+qKwcP4Ye2+JG=gGDCEPvRGgd+oDp-A@mail.gmail.com>

That option doesn't exist currently.  Take a look at the code, it shouldn't
be really hard to add one as long as not specifying it maintains current
behavior.  Give a try at a pull request to add the feature.

On Wednesday, February 24, 2016, Naveen Tamanam <naveen32india at gmail.com>
wrote:

> I would like to play different invalid sound for the first failure which
> would be long sound and some other invalid sound for next retries.
>
> Do we have any kind of option like greet-long and greet-short to play
> different invalid sounds, like invalid-long invalid-short ?
>
> --
> Thanks & Regards,
> Naveen Tamanam
>
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From mike at jerris.com  Wed Feb 24 16:22:32 2016
From: mike at jerris.com (Michael Jerris)
Date: Wed, 24 Feb 2016 08:22:32 -0500
Subject: [Freeswitch-users] libyuv problem
In-Reply-To: <1700116939.11698256.1456314900112.JavaMail.zimbra@panamaxil.com>
References: <1700116939.11698256.1456314900112.JavaMail.zimbra@panamaxil.com>
Message-ID: <CAEwTsAgb+Anvz-6OR5MTPx+AdA54r6Vi-yzoxPt3L6wVtSjpCA@mail.gmail.com>

It will only work if you use our libyuv package or tarballs. Fedoras
packages are broken, missing the pkg-config files.  Note we are in process
of moving this library in tree static to not use system libs along with vpx
due to needs to have much newer versions changing at a much higher rate
than system packages.  This change will come very soon.

On Wednesday, February 24, 2016, Veerabhadrarao Kankatala <
veerabhadrarao.kankatala at panamaxil.com> wrote:

> hello,
>
> I am getting following error while configrung freeswitch 1.6.5.
>
> ./configure (success)
>
> but when i run make command i am facing following error
>
> making all mod_fsv
> make[4]: Entering directory
> `/usr/src/bhadra/freeswitch-1.6.5/src/mod/applications/mod_fsv'
> Makefile:797: *** You must install libyuv-dev to build mod_fsv. Stop.
> make[4]: Leaving directory
> `/usr/src/bhadra/freeswitch-1.6.5/src/mod/applications/mod_fsv'
> make[3]: *** [mod_fsv-all] Error 1
> make[3]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5/src/mod'
> make[2]: *** [all-recursive] Error 1
> make[2]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5/src'
> make[1]: *** [all-recursive] Error 1
> make[1]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5'
> make: *** [all] Error 2
>
> but i already installed libyuv packages
>
> yum install libyuv-devel
> Loaded plugins: langpacks, presto, refresh-packagekit
> Package libyuv-devel-0-0.17.20121221svn522.fc18.x86_64 already installed
> and latest version
> Nothing to do
>
>
> i would like to know which version of libyuv, libyuv-devel  is require
> for freeswitch 1.6.5V
>
> I am using Fedora 18 64Bit,
>
> Please help me to come out from this problem
>
> Thanks in advance
>
> --
> Thanks & Regards
> *Veerabhadrarao Kankatala*
> Software Developer (C & Unix)
> *PANAMAX INFOTECH LIMITED*
> *Mobile:* *+91-8401231249*
> *Messenger Id:* Skype: veerabhadrarao.kankatala
> *E-mail: **veerabhadrarao.kankatala at panamaxil.com
> <javascript:_e(%7B%7D,'cvml','veerabhadrarao.kankatala at panamaxil.com');>*
>
>
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From mylists at polite.se  Wed Feb 24 18:23:50 2016
From: mylists at polite.se (Oivvio Polite)
Date: Wed, 24 Feb 2016 16:23:50 +0100
Subject: [Freeswitch-users] Joining several FS servers in different regions
Message-ID: <20160224152350.GA30297@blomma.liberationtech.net>


I'm working on a Saas service that will connect WebRTC (with SIP or Verto for
signaling) clients to one another and to PSTN. I'm aware of other such
services, but this is tailored to a specific niche. I also want WebRTC
media to flow through FS, rather than P2P, so that I can record it.

The primary markets are Europe and North America. To keep latency low I need
to have servers on both continents. Later on I might want to add more
regions. I'm not too worried about high availability or high loads. 

The clients can keep track of which FS is closest to them and register
with that. But this still leaves me with a couple of questions. 



1. Maintaining a single user database
=====================================

What's the simplest way of maintaining one joint user database? (User
directory in FS parlance). 

The first thing that comes to mind is to generate the xml dynamically
from a central database, and force as `reloadxml` on all servers every
time there's an update. The communication between central database and
FS servers could be via RabbitMQ.  mod_xml_curl could also be a part of
this I guess.

How does that sound? Is there an other simpler way of achiving my goal
that I'm not seeing?



2. Keeping track of where a user i currently registered
=======================================================

Let's say I have a client Alice who's registered with a FS server in
North America and a client Bob who has registered wit a FS server in Europe.

Now Alice tries to call Bob. How does the North America server know that
Bob is currently registered with the Europe server and that the call
should be routed through there?

For this second problem I don't even have a tentative solution so I'm
curious to hear any ideas.

3. SIP Trunks and geography
===========================

If Alice is in North America and wants to call a PSTN endpoint in North
America I should obviously be using a SIP trunk in North America, since
going over the Atlantic twice will add a lot of latency.

But what about when Alice want to call a PSTN endpoint in Europe? Is it
better to use a SIP endpoint in Europe or in North America?

Thanks in advance Oivvio


From ssinyagin at gmail.com  Wed Feb 24 18:49:01 2016
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Wed, 24 Feb 2016 16:49:01 +0100
Subject: [Freeswitch-users] Joining several FS servers in different
	regions
In-Reply-To: <20160224152350.GA30297@blomma.liberationtech.net>
References: <20160224152350.GA30297@blomma.liberationtech.net>
Message-ID: <CACOYK=gZ3GzJ-eNkSN9DhDnVPgkcXth2Oe9kTLeGmi82J56vEQ@mail.gmail.com>

On Wed, Feb 24, 2016 at 4:23 PM, Oivvio Polite <mylists at polite.se> wrote:
>
> I'm working on a Saas service that will connect WebRTC (with SIP or Verto for
> signaling) clients to one another and to PSTN. I'm aware of other such
> services, but this is tailored to a specific niche. I also want WebRTC
> media to flow through FS, rather than P2P, so that I can record it.
>
> The primary markets are Europe and North America. To keep latency low I need
> to have servers on both continents. Later on I might want to add more
> regions. I'm not too worried about high availability or high loads.
>
> The clients can keep track of which FS is closest to them and register
> with that. But this still leaves me with a couple of questions.
>
>
>
> 1. Maintaining a single user database
> =====================================
>
> What's the simplest way of maintaining one joint user database? (User
> directory in FS parlance).
>
> The first thing that comes to mind is to generate the xml dynamically
> from a central database, and force as `reloadxml` on all servers every
> time there's an update. The communication between central database and
> FS servers could be via RabbitMQ.  mod_xml_curl could also be a part of
> this I guess.
>
> How does that sound? Is there an other simpler way of achiving my goal
> that I'm not seeing?
>

You can use mod_xml_curl for this. Every time FS receives a REGISTER
message from the user, it would retrieve a piece of XML from your
application HTTP service. As these requests need to be answered quite
quickly, you would rather install the HTTP server in the same location
as FreeSWITCH. Also keep in mind that every call to "user_data"
function would trigger an extra HTTP request for the user's XML. The
function is quite convenient if you want to get some user-specific
parameters from the directory.

Then it's up to you how you synchronize the back-end databases.
Probably something like https://ipfs.io/  would help.



>
>
> 2. Keeping track of where a user i currently registered
> =======================================================
>
> Let's say I have a client Alice who's registered with a FS server in
> North America and a client Bob who has registered wit a FS server in Europe.
>
> Now Alice tries to call Bob. How does the North America server know that
> Bob is currently registered with the Europe server and that the call
> should be routed through there?
>
> For this second problem I don't even have a tentative solution so I'm
> curious to hear any ideas.

Each successful registration generates an event (if I'm not mistaken),
and an entry in FreeSWITCH's own database. So, you should be able to
retrieve the IP address of the FreeSWITCH server where the user got
registered.

Alternatively, you may have one Kamailio cluster in one location of
the world, acting as a SIP registrar. A 250ms round-trip delay is not
so critical for SIP traffic.

Alternatively, you can have one FreeSWITCH cluster acting as a SIP
registrar and call routing engine in media pass-through mode. Then few
other FreeSWITCH servers would work as media gateways in different
regions.


> 3. SIP Trunks and geography
> ===========================
>
> If Alice is in North America and wants to call a PSTN endpoint in North
> America I should obviously be using a SIP trunk in North America, since
> going over the Atlantic twice will add a lot of latency.
>
> But what about when Alice want to call a PSTN endpoint in Europe? Is it
> better to use a SIP endpoint in Europe or in North America?
>

This needs testing. I would expect that global VoIP carriers like
Voxbone have better control on their bandwidth and transport networks.


From dominique.jeannerod at interact-iv.com  Wed Feb 24 19:38:28 2016
From: dominique.jeannerod at interact-iv.com (Dominique Jeannerod)
Date: Wed, 24 Feb 2016 17:38:28 +0100
Subject: [Freeswitch-users] mod_distributor / gwlist down
Message-ID: <CAEzwe14uLCk+MvyQ8mNuixR1Gy13D8ox+TKbOTt9ucQEWdePsw@mail.gmail.com>

Hello,

i'm currently working with mod_distributor, to load balance calls to
gateways.
The load balancing works well, and it's rather simple to configure, no
problem.

I'm trying to automatically exclude down gateways from the load balancing,
which seems quite simple, too, when reading the mod_distributor doc... but
i can't make it work.

I'm using freeswitch 1.6.6.
I configured a profile internal_trk_1, and 2 gateways (svc-isr-fr-301, and
non_existent_gw), with ping option.
One of the gateways, called (non_existent_gw) is down, and detected down by
sofia

When I check the status of the gateways, the result is ok :
sofia profile internal_trk_1 gwlist down
non_existent_gw

When I check with distributor command in the cli, it's also ok :
expand distributor I_MUT_SIPGTW ${sofia profile internal_trk_1 gwlist down}
svc-isr-fr-301

But I can't make it work inside the dialplan. I tested using the 2 syntaxes
listed in
https://freeswitch.org/confluence/display/FREESWITCH/mod_distributor :

<action application="bridge" data="sofia/gateway/${distributor I_MUT_SIPGTW
${sofia internal_trk_1 gwlist down}}/${destination_number}"/>

<action application="bridge" data="sofia/gateway\${expand(distributor
I_MUT_SIPGTW \${sofia(internal_trk_1 gwlist
down)})}/${destination_number}"/>

The dead gateway still gets used by the load balancer.

What I am doing wrong ?

Best regards


Dominique Jeannerod
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From mike at jerris.com  Thu Feb 25 02:49:31 2016
From: mike at jerris.com (Michael Jerris)
Date: Wed, 24 Feb 2016 18:49:31 -0500
Subject: [Freeswitch-users] major build change in master.
Message-ID: <F7862F99-22B2-4165-8EE3-9872E848B14E@jerris.com>

please note this important change that went into master today.  As of today, we are no longer using system versions of libyuv and libvpx due to major conflicts with system versions of these libraries.  These are now built static into the freeswitch core.  Also note, mod_vpx no longer exists, it is automatically loaded as part of the core and you will no longer have mod_vpx.so or have to manually load it.  I'll have more details coming, but let me know if you have any questions.  Please note for anyone doing cross compiling, this probably needs a bit more work for you, I'll be working on fixing that this week.

Mike

From Shawn.Wheeler at interlockconcepts.com  Tue Feb 23 22:24:16 2016
From: Shawn.Wheeler at interlockconcepts.com (Shawn Wheeler)
Date: Tue, 23 Feb 2016 19:24:16 +0000
Subject: [Freeswitch-users] NUBE to FreeSWITCH  - Windows?
Message-ID: <SN1PR0201MB15042F81BDE2AE1D55E12A84E0A40@SN1PR0201MB1504.namprd02.prod.outlook.com>

I know there is a windows installer for FreeSWITCH.  Does anyone know if this will work with Win 7 32bit?  It is what I have a physical Test box

Thank you in advance

Shawn
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From afarooqa at gmail.com  Wed Feb 24 01:51:37 2016
From: afarooqa at gmail.com (Faruq Ahmad)
Date: Wed, 24 Feb 2016 03:51:37 +0500
Subject: [Freeswitch-users] mod_sms chatplan default behaviour
Message-ID: <CAM=x0DOFZyzoFg81gBBPZRgf_p8x4X81WePGy5cueh6J7k6cew@mail.gmail.com>

Hi,

working on mod_sms,
using the demo chatplan called a python script, the script generates a
reply for the sms and sends it using chat_execute reply application.
however once the script has completed the sms is forwarded to the
extensions in "to" field as well. I dont see any application in chatplan
that is forwarding the message to the extension. can this feature be
controlled? for example if the script called in chatplan has sent a reply
for the sms, it is not longer sent to the destination extension?

also When sending a message using ESL, logs only show "SMS Delivery assumed
successful due to being sent in non-blocking manner" if the destination
extension is available. if the destination extensions is not available I
get "Nobody to send to" as well as the above message. how can I get any
more details for these events. script that is used to send sms has

f = con.sendEvent(event)

print f.serialize() always shows the following no matter if the message is
sent ahead or not.

Event-Name: SOCKET_DATA
Content-Type: command/reply
Reply-Text: %2BOK%20accepted
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From freeswitch at digitaldescent.net  Thu Feb 25 06:00:06 2016
From: freeswitch at digitaldescent.net (Brian Chow)
Date: Wed, 24 Feb 2016 19:00:06 -0800
Subject: [Freeswitch-users] Inbound calls mis-routing routing to internal
	extension with external IP?
Message-ID: <CAA_Md=uAGmnqLkY3g128QcjoBis4VnoTwSUEZpqAws35AwRsHQ@mail.gmail.com>

Hello all!
  I'm new to freeswitch, so I'm sure this is just a newbie configuration
error.  Sorry if it's been answered a million times, my google searches
always just bring up the standard NAT configuration pages.  I've already
followed the confluence page on configuring NAT.

My Setup:
Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3 instance.
SIP Provider: Flowroute

NAT: extension and freeswitch are on the same network, both of which are
behind NAT and connecting to flowroute.  I configured external sip and rtp
to use stun entries.  External profile is using $${external_rtp/sip_ip} for
ext-rtp/sip respectively.

I have one DID configured to go directly to my one extension.  My extension
can register just fine.  My extension can dial out.  When I call my cell,
the call connects and I have 2 way audio.  When I dial my DID from my cell,
I can see the call hitting the FS server, but instead of ringing my
extension, it goes straight to my extensions voicemail (which I can just
fine).

When I look the at the console, it appears (sorry if this is wrong, I'm
only a day into free switch) that FS is attempting to route the call to my
extension...@ my external ip?

I see this line in the console:  2016-02-24 18:56:10.453983 [NOTICE]
switch_channel.c:1101 New Channel sofia/internal/<extension>@*<externalip>*
:46072


Shouldn't that read sofia/internal/<extentsion>@<ip_of_extension>  ?

Sofia Status says:

                     Name   Type                                      Data
State
=================================================================================================
            external-ipv6 profile
sip:mod_sofia@[::1]:5080 RUNNING
(0)
                 external profile
sip:mod_sofia@<external_ip>:5080 RUNNING
(0)
external::<uuid> gateway            sip:<flowrouteid>@sip.flowroute.com
REGED
            internal-ipv6 profile
sip:mod_sofia@[::1]:5060 RUNNING
(0)
                 internal profile
sip:mod_sofia@<internal_ip>:5060 RUNNING
(0)
=================================================================================================

If I'm completely off base here, can anyone recommend where I can start
looking to change/troubleshoot the issue?  I feel like it's just me missing
something, I just can't determine what that might be.

Thanks,
  -Brian
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From krice at freeswitch.org  Thu Feb 25 06:17:28 2016
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 24 Feb 2016 21:17:28 -0600
Subject: [Freeswitch-users] NUBE to FreeSWITCH  - Windows?
In-Reply-To: <SN1PR0201MB15042F81BDE2AE1D55E12A84E0A40@SN1PR0201MB1504.namprd02.prod.outlook.com>
References: <SN1PR0201MB15042F81BDE2AE1D55E12A84E0A40@SN1PR0201MB1504.namprd02.prod.outlook.com>
Message-ID: <32ae01d16f7b$0f52f310$2df8d930$@freeswitch.org>

There is a newish 64bit installer on files.freeswitch.org but I doubt theres an recent 32 bit one

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shawn Wheeler
Sent: Tuesday, February 23, 2016 1:24 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] NUBE to FreeSWITCH - Windows?

 

I know there is a windows installer for FreeSWITCH.  Does anyone know if this will work with Win 7 32bit?  It is what I have a physical Test box

 

Thank you in advance

 

Shawn

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From max at nysolutions.com  Thu Feb 25 07:15:36 2016
From: max at nysolutions.com (Moishe Grunstein)
Date: Thu, 25 Feb 2016 04:15:36 +0000
Subject: [Freeswitch-users] NUBE to FreeSWITCH  - Windows?
In-Reply-To: <32ae01d16f7b$0f52f310$2df8d930$@freeswitch.org>
References: <SN1PR0201MB15042F81BDE2AE1D55E12A84E0A40@SN1PR0201MB1504.namprd02.prod.outlook.com>
	<32ae01d16f7b$0f52f310$2df8d930$@freeswitch.org>
Message-ID: <df4372e9e4fc4b34a5d8c15bf39172d8@nysolutions.com>

Win32 installer over a year old http://files.freeswitch.org/windows_installer/installer/x86/freeswitch.msi
Win64 installer just over 1 month old http://files.freeswitch.org/windows_installer/installer/x64/freeswitch.msi


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 Ken Rice
Sent: Wednesday, February 24, 2016 10:17 PM
To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] NUBE to FreeSWITCH - Windows?

There is a newish 64bit installer on files.freeswitch.org but I doubt theres an recent 32 bit one

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 Shawn Wheeler
Sent: Tuesday, February 23, 2016 1:24 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: [Freeswitch-users] NUBE to FreeSWITCH - Windows?

I know there is a windows installer for FreeSWITCH.  Does anyone know if this will work with Win 7 32bit?  It is what I have a physical Test box

Thank you in advance

Shawn
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From max at nysolutions.com  Thu Feb 25 07:31:37 2016
From: max at nysolutions.com (Moishe Grunstein)
Date: Thu, 25 Feb 2016 04:31:37 +0000
Subject: [Freeswitch-users] Inbound calls mis-routing routing to
	internal	extension with external IP?
In-Reply-To: <CAA_Md=uAGmnqLkY3g128QcjoBis4VnoTwSUEZpqAws35AwRsHQ@mail.gmail.com>
References: <CAA_Md=uAGmnqLkY3g128QcjoBis4VnoTwSUEZpqAws35AwRsHQ@mail.gmail.com>
Message-ID: <23b1626b3814448b90842a279dc30be5@nysolutions.com>

It is probably sending call to your extension at domainname, if your external ip is your domain name then you will see the external ip.
Is your endpoint registered? How often is it set to reregister?
Does your router have a sip alg? Are the ports opened in your firewall?
Very hard to guess without a log of a call with debug enabled.
sofia global sip trace on

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 Brian Chow
Sent: Wednesday, February 24, 2016 10:00 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP?

Hello all!
  I'm new to freeswitch, so I'm sure this is just a newbie configuration error.  Sorry if it's been answered a million times, my google searches always just bring up the standard NAT configuration pages.  I've already followed the confluence page on configuring NAT.

My Setup:
Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3 instance.
SIP Provider: Flowroute

NAT: extension and freeswitch are on the same network, both of which are behind NAT and connecting to flowroute.  I configured external sip and rtp to use stun entries.  External profile is using $${external_rtp/sip_ip} for ext-rtp/sip respectively.

I have one DID configured to go directly to my one extension.  My extension can register just fine.  My extension can dial out.  When I call my cell, the call connects and I have 2 way audio.  When I dial my DID from my cell, I can see the call hitting the FS server, but instead of ringing my extension, it goes straight to my extensions voicemail (which I can just fine).

When I look the at the console, it appears (sorry if this is wrong, I'm only a day into free switch) that FS is attempting to route the call to my extension...@ my external ip?

I see this line in the console:  2016-02-24 18:56:10.453983 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/<extension>@<externalip>:46072


Shouldn't that read sofia/internal/<extentsion>@<ip_of_extension>  ?

Sofia Status says:

                     Name        Type                                      Data  State
=================================================================================================
            external-ipv6   profile                   sip:mod_sofia@[::1]:5080            RUNNING (0)
                 external      profile           sip:mod_sofia@<external_ip>:5080<sip:mod_sofia@%3cexternal_ip%3e:5080>     RUNNING (0)
external::<uuid>          gateway                      sip:<flowrouteid>@sip.flowroute.com<http://sip.flowroute.com>          REGED
            internal-ipv6    profile                   sip:mod_sofia@[::1]:5060            RUNNING (0)
                 internal       profile            sip:mod_sofia@<internal_ip>:5060<sip:mod_sofia@%3cinternal_ip%3e:5060>    RUNNING (0)
=================================================================================================

If I'm completely off base here, can anyone recommend where I can start looking to change/troubleshoot the issue?  I feel like it's just me missing something, I just can't determine what that might be.

Thanks,
  -Brian
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From Alexander.Haugg at c4b.de  Thu Feb 25 09:58:49 2016
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Thu, 25 Feb 2016 06:58:49 +0000
Subject: [Freeswitch-users] Web RTC -> No Candidate found
In-Reply-To: <CAEJMVkAn26kvUtFMtq+U+kPsTEjR-3FHxD1mFADXUqsZuw=gww@mail.gmail.com>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC8C0059F58@vc4bex2.corp.c4b.de>
	<CAEJMVkAn26kvUtFMtq+U+kPsTEjR-3FHxD1mFADXUqsZuw=gww@mail.gmail.com>
Message-ID: <B60DA91F5EA0E24A96E73CDE43CC9EC8C005A731@vc4bex2.corp.c4b.de>

This works, thanks

Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Brian West
Gesendet: Donnerstag, 18. Februar 2016 15:31
An: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Betreff: Re: [Freeswitch-users] Web RTC -> No Candidate found

Try
apply-candidate-acl

On Thu, Feb 18, 2016 at 2:29 AM, Alexander Haugg <Alexander.Haugg at c4b.de<mailto:Alexander.Haugg at c4b.de>> wrote:
Hi,

I have the problem, that the freeswitch don?t accept the network internal candidate.

The local IP address of the fs is 192.168.241.5
The local IP address of the webrtc Client is 192.168.241.2

In the acl.conf.xml is set:
<list name="localnet.auto" default="allow">
                <node type="allow" cidr="192.168.241.0/24<http://192.168.241.0/24>"/>
</list>

In the sofia profile is set:
<param name="local-network-acl" value="localnet.auto"/>
<param name="apply-inbound-acl" value="localnet.auto"/>

The following SDP is send by the client:
v=0
o=- 222075295642546336 815544600 IN IP4 127.0.0.1
s=IceLink
t=0 0
m=audio 30450 RTP/SAVPF 0 8
c=IN IP4 192.168.62.1
a=rtcp:30450 IN IP4 192.168.62.1
a=ice-ufrag:532479e3
a=ice-pwd:4b066f2e64633d218085fc39c49a443c
a=sendrecv
a=rtcp-mux
a=fingerprint:sha-256 FD:DE:DB:44:C8:66:10:B0:DF:3A:D5:17:4D:89:0E:1A:39:0C:99:D3:AE:BE:1F:F2:D7:78:2E:8E:A6:6B:F1:36
a=setup:actpass
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ssrc:967296980 cname:c2088983
a=candidate:954dc3ed263236386e8615555070c83dg 1 udp 2130706331 192.168.241.2 44376 typ host
a=candidate:954dc3ed263236386e8615555070c83dg 2 udp 2130706331 192.168.241.2 44376 typ host

The output on the fs_cli:
2016-02-18 07:36:33.568868 [DEBUG] switch_channel.c:3759 (sofia/H3KSip/170) Callstate Change RINGING -> ACTIVE
2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4208 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4208 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:2898 Set Codec sofia/H3KSip_B2Bua/170 at 192.168.241.2:5058<http://170 at 192.168.241.2:5058> PCMU/8000 20 ms 160 samples 64000 bits 1 channels
2016-02-18 07:36:33.588870 [DEBUG] switch_core_codec.c:111 sofia/H3KSip_B2Bua/170 at 192.168.241.2:5058<http://170 at 192.168.241.2:5058> Original read codec set to PCMU:0
2016-02-18 07:36:33.588870 [WARNING] switch_core_media.c:3266 NO candidate ACL defined, Defaulting to wan.auto
2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3296 Save audio Candidate cid: 1 proto: udp type: host addr: 192.168.241.2:44376<http://192.168.241.2:44376>
2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3296 Save audio Candidate cid: 2 proto: udp type: host addr: 192.168.241.2:443760<http://192.168.241.2:443760>
2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3336 Searching for rtp candidate.
2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3336 Searching for rtcp candidate.
2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3380 sofia/H3KSip_B2Bua/170 at 192.168.241.2:5058<http://170 at 192.168.241.2:5058> no suitable candidates found.
2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4480 No 2833 in SDP.  Disable 2833 dtmf and switch to INFO
2016-02-18 07:36:33.588870 [NOTICE] switch_channel.c:3798 Hangup sofia/H3KSip_B2Bua/170 at 192.168.241.2:5058<http://170 at 192.168.241.2:5058> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION

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



--

Brian West
brian at freeswitch.org<mailto:brian at freeswitch.org>

[http://billing.freeswitch.org/templates/default/img/whmcslogo.png]

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

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 findmeinwland at gmail.com  Thu Feb 25 11:19:44 2016
From: findmeinwland at gmail.com (Artur Mega)
Date: Thu, 25 Feb 2016 13:19:44 +0500
Subject: [Freeswitch-users] serving configuration with mod_lua,
	and sofia.conf
Message-ID: <CAOLEQ7dJ0ptSt3TqS16L7Ue_V_oAeQb5Jd43t=S8PXZOYkQ8jA@mail.gmail.com>

Hello all,

I try to serve configuration via lua like it's described at
https://freeswitch.org/confluence/display/FREESWITCH/Serving+Configuration+with+Lua

<configuration name="lua.conf" description="LUA Configuration">
 <settings>
   <param name="xml-handler-script" value="configuration.lua"/>
   <param name="xml-handler-bindings" value="configuration"/>
 </settings>
</configuration>

In configuration.lua script I set XML_STRING to valid xml, taken from
autoload_configs.

Similar to this method, i did the same with mod_xml_curl, and all works
fine.

But with mod_lua sofia profiles not loading. I think, the problem is in
variables. Let me explain.
When I turn off mod_lua and then make `reload mod_sofia` in fs_cli i see
something like
...
2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 context [internal]
2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 rfc2833-pt [101]
2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 sip-port [5060]
2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 dialplan [XML]
2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 dtmf-duration [2000]
2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 inbound-codec-prefs
[OPUS,G722,PCMU,PCMA,VP8]
2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 outbound-codec-prefs
[OPUS,G722,PCMU,PCMA,VP8]
2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 rtp-timer-name [soft]
2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 rtp-ip [xxx.xx.x.x]
....

But when i turn on mod_lua and relod mod_sofia, in logs i see, that
variables not calculated!

2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 context [internal]
2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 rfc2833-pt [101]
2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 sip-port
[$${internal_sip_port}]
2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 dialplan [XML]
2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 dtmf-duration [2000]
2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 inbound-codec-prefs
[$${global_codec_prefs}]
2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 outbound-codec-prefs
[$${global_codec_prefs}]
2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 rtp-timer-name [soft]
2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 rtp-ip [$${local_ip_v4}]

If i run in fs_cli `eval $${local_ip_v4}`, it prints calculated IP. Maybe I
missed something? Thanks

-- 

Arthur
??
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From mylists at polite.se  Thu Feb 25 13:12:13 2016
From: mylists at polite.se (Oivvio Polite)
Date: Thu, 25 Feb 2016 11:12:13 +0100
Subject: [Freeswitch-users] Joining several FS servers in different
 regions
In-Reply-To: <CACOYK=gZ3GzJ-eNkSN9DhDnVPgkcXth2Oe9kTLeGmi82J56vEQ@mail.gmail.com>
References: <20160224152350.GA30297@blomma.liberationtech.net>
	<CACOYK=gZ3GzJ-eNkSN9DhDnVPgkcXth2Oe9kTLeGmi82J56vEQ@mail.gmail.com>
Message-ID: <20160225101213.GB30297@blomma.liberationtech.net>



On ons, feb 24, 2016 at 04:49:01 +0100, Stanislav Sinyagin wrote:
> You can use mod_xml_curl for this. Every time FS receives a REGISTER
> message from the user, it would retrieve a piece of XML from your
> application HTTP service. As these requests need to be answered quite
> quickly, you would rather install the HTTP server in the same location
> as FreeSWITCH. Also keep in mind that every call to "user_data"
> function would trigger an extra HTTP request for the user's XML. The
> function is quite convenient if you want to get some user-specific
> parameters from the directory.
> 
> Then it's up to you how you synchronize the back-end databases.
> Probably something like https://ipfs.io/  would help.
> 
> 
> 
> Each successful registration generates an event (if I'm not mistaken),
> and an entry in FreeSWITCH's own database. So, you should be able to
> retrieve the IP address of the FreeSWITCH server where the user got
> registered.
> 
> Alternatively, you may have one Kamailio cluster in one location of
> the world, acting as a SIP registrar. A 250ms round-trip delay is not
> so critical for SIP traffic.
> 
> Alternatively, you can have one FreeSWITCH cluster acting as a SIP
> registrar and call routing engine in media pass-through mode. Then few
> other FreeSWITCH servers would work as media gateways in different
> regions.
> 
> 
> 
> This needs testing. I would expect that global VoIP carriers like
> Voxbone have better control on their bandwidth and transport networks.
> 

Thanks Stanislav for your suggestions. I think the idea with a Kamailio
cluster (or in my case a single box) might be a good fit for my
scenario. I'll go read up on that.

regards Oivvio


From gregor at infomedia.si  Thu Feb 25 14:05:45 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Thu, 25 Feb 2016 12:05:45 +0100
Subject: [Freeswitch-users] mod_amqp for Windows
Message-ID: <CAC7ttFEPYwit2r5XYgY31EQgxuN5bD_sLUP6WzzfAbJONYGF_g@mail.gmail.com>

Hi!

Does anybody knows if module mod_amqp  is available for windows if I build
by myself?

Best regards, Gregor
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From gb at cm.nl  Thu Feb 25 14:20:54 2016
From: gb at cm.nl (Grant Bagdasarian)
Date: Thu, 25 Feb 2016 11:20:54 +0000
Subject: [Freeswitch-users] Difference between mod_avmd and mod_com_amd
Message-ID: <8168d1e51be74b4f864a572ebeca6f25@CM-EX-V01.cm.local>

Hello,

Could someone explain to me the difference between the modules mod_avmd and mod_com_amd, except for the price different?

Regards,

Grant
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From italo at freeswitch.org  Thu Feb 25 14:49:57 2016
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Thu, 25 Feb 2016 08:49:57 -0300
Subject: [Freeswitch-users] Difference between mod_avmd and mod_com_amd
In-Reply-To: <8168d1e51be74b4f864a572ebeca6f25@CM-EX-V01.cm.local>
References: <8168d1e51be74b4f864a572ebeca6f25@CM-EX-V01.cm.local>
Message-ID: <CABMX_w9=_eve+wWTE8g-d9hZNEBsoVn6zvq9MD_77g7LfrZfBg@mail.gmail.com>

mod_avmd can detect the presence of BEEPS on a call and mod_com_amd can
analyze patterns of silence and voice and determine if there's an human or
a machine in the remote side.

On Thu, Feb 25, 2016 at 8:20 AM, Grant Bagdasarian <gb at cm.nl> wrote:

> Hello,
>
>
>
> Could someone explain to me the difference between the modules mod_avmd
> and mod_com_amd, except for the price different?
>
>
>
> 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
>



-- 
?talo Rossi
italo at freeswitch.org
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From italo at freeswitch.org  Thu Feb 25 14:55:02 2016
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Thu, 25 Feb 2016 08:55:02 -0300
Subject: [Freeswitch-users] gsmopen problem: INCOMPATIBLE_DESTINATION
In-Reply-To: <CAHKqMb7MosEn_kA5KYhME+rfG5G+3Yqwiuoexanww=CFZb+N8g@mail.gmail.com>
References: <CAHKqMb7MosEn_kA5KYhME+rfG5G+3Yqwiuoexanww=CFZb+N8g@mail.gmail.com>
Message-ID: <CABMX_w-nfvq2aeF8VhXZmQGi-PjLURnXtrHT55_XAV32XVCzrQ@mail.gmail.com>

What version are you using? Try setting
media_mix_inbound_outbound_codecs=true

On Tue, Feb 23, 2016 at 12:23 AM, Nandy Dagondon <nandy1925 at gmail.com>
wrote:

> Hello guys,
>
> I installed mod_gsmopen. When testing incoming call, it hangs up with this
> error message INCOMPATIBLE_DESTINATION.  I understand this is a codec
> negotiation issue.
>
> Looking at the CDR - the gsmopen presented L16 8000 in the SDP.  By
> default, FS does early negotiation.
>
> Snippet of my internal sip_profile:
> <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="presence-privacy" value="$${presence_privacy}"/>
> <param name="inbound-codec-negotiation" value="generous"/>
>
> Take note: there is no disable-transcoding parameter which I understand FS
> is transcodind.
>
> In vars.xml:
>
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h
> ,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>
> I tried to add L16 at the end of global_codec_prefs which is unnecessary,
> isn't it? But it didn't work, too.
>
> Finally, I added L16 in the softphone, it worked.
>
> I have installed gsmopen before and it worked (transcoding) with IP
> phones.
>
> Is there anything I may have missed?  I appreciate for any leads.
>
> Thank you,
>
> /Nandy
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From gb at cm.nl  Thu Feb 25 15:07:49 2016
From: gb at cm.nl (Grant Bagdasarian)
Date: Thu, 25 Feb 2016 12:07:49 +0000
Subject: [Freeswitch-users] Difference between mod_avmd and mod_com_amd
In-Reply-To: <CABMX_w9=_eve+wWTE8g-d9hZNEBsoVn6zvq9MD_77g7LfrZfBg@mail.gmail.com>
References: <8168d1e51be74b4f864a572ebeca6f25@CM-EX-V01.cm.local>
	<CABMX_w9=_eve+wWTE8g-d9hZNEBsoVn6zvq9MD_77g7LfrZfBg@mail.gmail.com>
Message-ID: <0604e122c6614e549b8b33094acf4b7a@CM-EX-V01.cm.local>

Thanks!

Do both modules give you an event or variable value after ?successful? machine detection, be it using event socket or plain XML?

Regards,

Grant

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of ?talo Rossi
Sent: Thursday, February 25, 2016 12:50 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Difference between mod_avmd and mod_com_amd

mod_avmd can detect the presence of BEEPS on a call and mod_com_amd can analyze patterns of silence and voice and determine if there's an human or a machine in the remote side.

On Thu, Feb 25, 2016 at 8:20 AM, Grant Bagdasarian <gb at cm.nl<mailto:gb at cm.nl>> wrote:
Hello,

Could someone explain to me the difference between the modules mod_avmd and mod_com_amd, except for the price different?

Regards,

Grant

_________________________________________________________________________
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



--
?talo Rossi
italo at freeswitch.org<mailto:italo at freeswitch.org>
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From italo at freeswitch.org  Thu Feb 25 15:11:57 2016
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Thu, 25 Feb 2016 09:11:57 -0300
Subject: [Freeswitch-users] Difference between mod_avmd and mod_com_amd
In-Reply-To: <0604e122c6614e549b8b33094acf4b7a@CM-EX-V01.cm.local>
References: <8168d1e51be74b4f864a572ebeca6f25@CM-EX-V01.cm.local>
	<CABMX_w9=_eve+wWTE8g-d9hZNEBsoVn6zvq9MD_77g7LfrZfBg@mail.gmail.com>
	<0604e122c6614e549b8b33094acf4b7a@CM-EX-V01.cm.local>
Message-ID: <CABMX_w9re70W7vYnQ-SSJ7YT6AgAWmc38B=vuE=xoig0kSGbOg@mail.gmail.com>

Yes, for both.

https://freeswitch.org/confluence/display/FREESWITCH/mod_com_amd
https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd

Read the examples.

On Thu, Feb 25, 2016 at 9:07 AM, Grant Bagdasarian <gb at cm.nl> wrote:

> Thanks!
>
>
>
> Do both modules give you an event or variable value after ?successful?
> machine detection, be it using event socket or plain XML?
>
>
>
> Regards,
>
>
>
> Grant
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *?talo Rossi
> *Sent:* Thursday, February 25, 2016 12:50 PM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Difference between mod_avmd and
> mod_com_amd
>
>
>
> mod_avmd can detect the presence of BEEPS on a call and mod_com_amd can
> analyze patterns of silence and voice and determine if there's an human or
> a machine in the remote side.
>
>
>
> On Thu, Feb 25, 2016 at 8:20 AM, Grant Bagdasarian <gb at cm.nl> wrote:
>
> Hello,
>
>
>
> Could someone explain to me the difference between the modules mod_avmd
> and mod_com_amd, except for the price different?
>
>
>
> 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
>
>
>
>
>
> --
>
> ?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
>



-- 
?talo Rossi
italo at freeswitch.org
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From freeswitch at digitaldescent.net  Thu Feb 25 08:33:06 2016
From: freeswitch at digitaldescent.net (Brian Chow)
Date: Wed, 24 Feb 2016 21:33:06 -0800
Subject: [Freeswitch-users] Inbound calls mis-routing routing to
 internal extension with external IP?
In-Reply-To: <23b1626b3814448b90842a279dc30be5@nysolutions.com>
References: <CAA_Md=uAGmnqLkY3g128QcjoBis4VnoTwSUEZpqAws35AwRsHQ@mail.gmail.com>
	<23b1626b3814448b90842a279dc30be5@nysolutions.com>
Message-ID: <CAA_Md=t4NR-ocAsnvOmD39XGNAexYVZWsWy4taYbiXU7-N1qEg@mail.gmail.com>

Debug console output: https://pastebin.freeswitch.org/24572

My fqdn only resolves inside the network, but the external-ip is configured
in FS to use stun, which reports my external ip.

>Is your endpoint registered? How often is it set to reregister?

Yes, and the zoiper default registration expiry is 3600



>Does your router have a sip alg? Are the ports opened in your firewall?

> ish, no GUI for it, but I did make sure to disable it from the command
line (Ubiquiti Edgerouter).  No firewall ports are explicitly open, but
allow bi-directional with existing states.  I do get 2 way audio when
calling out, and I also can hear the system forwarding my call in to
voicemail; the voicemail message and prompts.  I know asterisk is different
from FS, but my asterisk experience has proven that at least with asterisk,
port forwarding isn't needed.  Also, unfortunately, I cannot get a static
ip.


Thanks!

  -Brian

On Wed, Feb 24, 2016 at 8:31 PM, Moishe Grunstein <max at nysolutions.com>
wrote:

> It is probably sending call to your extension at domainname, if your
> external ip is your domain name then you will see the external ip.
>
> Is your endpoint registered? How often is it set to reregister?
>
> Does your router have a sip alg? Are the ports opened in your firewall?
>
> Very hard to guess without a log of a call with debug enabled.
>
> sofia global sip trace on
>
>
>
> 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 *Brian Chow
> *Sent:* Wednesday, February 24, 2016 10:00 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Inbound calls mis-routing routing to
> internal extension with external IP?
>
>
>
> Hello all!
>
>   I'm new to freeswitch, so I'm sure this is just a newbie configuration
> error.  Sorry if it's been answered a million times, my google searches
> always just bring up the standard NAT configuration pages.  I've already
> followed the confluence page on configuring NAT.
>
>
>
> My Setup:
>
> Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3
> instance.
>
> SIP Provider: Flowroute
>
>
>
> NAT: extension and freeswitch are on the same network, both of which are
> behind NAT and connecting to flowroute.  I configured external sip and rtp
> to use stun entries.  External profile is using $${external_rtp/sip_ip} for
> ext-rtp/sip respectively.
>
>
>
> I have one DID configured to go directly to my one extension.  My
> extension can register just fine.  My extension can dial out.  When I call
> my cell, the call connects and I have 2 way audio.  When I dial my DID from
> my cell, I can see the call hitting the FS server, but instead of ringing
> my extension, it goes straight to my extensions voicemail (which I can just
> fine).
>
>
>
> When I look the at the console, it appears (sorry if this is wrong, I'm
> only a day into free switch) that FS is attempting to route the call to my
> extension...@ my external ip?
>
>
>
> I see this line in the console:  2016-02-24 18:56:10.453983 [NOTICE]
> switch_channel.c:1101 New Channel sofia/internal/<extension>@
> *<externalip>*:46072
>
>
>
>
>
> Shouldn't that read sofia/internal/<extentsion>@<ip_of_extension>  ?
>
>
>
> Sofia Status says:
>
>
>
>                      Name        Type
>  Data  State
>
>
> =================================================================================================
>
>             external-ipv6   profile
> sip:mod_sofia@[::1]:5080            RUNNING (0)
>
>                  external      profile
> sip:mod_sofia@<external_ip>:5080     RUNNING (0)
>
> external::<uuid>          gateway                      sip:<flowrouteid>@
> sip.flowroute.com          REGED
>
>             internal-ipv6    profile
> sip:mod_sofia@[::1]:5060            RUNNING (0)
>
>                  internal       profile
> sip:mod_sofia@<internal_ip>:5060    RUNNING (0)
>
>
> =================================================================================================
>
>
>
> If I'm completely off base here, can anyone recommend where I can start
> looking to change/troubleshoot the issue?  I feel like it's just me missing
> something, I just can't determine what that might be.
>
>
>
> Thanks,
>
>   -Brian
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 freeswitch.opencode at spamgourmet.com  Thu Feb 25 10:35:57 2016
From: freeswitch.opencode at spamgourmet.com (freeswitch.opencode at spamgourmet.com)
Date: Thu, 25 Feb 2016 07:35:57 +0000
Subject: [Freeswitch-users] Getting 503 when starting automatically via
	systemd
Message-ID: <BN3PR13MB0785EEACC5A82AAF57D9B2F1D7A60@BN3PR13MB0785.namprd13.prod.outlook.com>

I'm using Arch Linux (specifically Arch Linux ARM on a Raspberry Pi Zero). I'm successfully done a make and install for freeswitch 1.6.6 and when I run it manually (freeswitch -nonat), it works fine. Also, when I start it manually via systemd (systemctl start freeswitch), it still works fine.

However, when I enable freeswitch.service to start automatically and reboot, the freeswitch log shows 503 errors trying to register out to my gateway, and my internal client (an Obi200) gets a 503 when trying to register in to FreeSWITCH as well; this keeps happening repeatedly (waiting 30 seconds and then failing again). However, if I restart the service (systemctl restart freeswitch), connectivity then starts working.

Would the 503 be what's reported up if there's no network connectivity either way?

I'm using the Debian systemd *freeswitch.service file (with very minor tweaks for my configuration on Arch Linux). Has anyone encountered this problem before (on Debian, Arch Linux, or elsewhere)? The overall .service file is very similar to sshd.service, which doesn't have connection problems, so my guess is there's something specific to FreeSWITCH going on. If this is Arch Linux-specific and I ask in their forums, is there anything different about how FreeSWITCH handles connectivity (perhaps caching some kind of handle) that would help point things in the right direction?

Thanks,

David


From alessandro.illiano at toctoc.me  Thu Feb 25 14:15:03 2016
From: alessandro.illiano at toctoc.me (Alessandro Illiano)
Date: Thu, 25 Feb 2016 12:15:03 +0100
Subject: [Freeswitch-users] mod_rtmp rtmps support
Message-ID: <56CEE237.6060108@toctoc.me>

Hi all,
does mod_rtmp support rtmps?
Thanks

Alessandro


From brian at freeswitch.org  Thu Feb 25 16:50:37 2016
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Feb 2016 07:50:37 -0600
Subject: [Freeswitch-users] Inbound calls mis-routing routing to
 internal extension with external IP?
In-Reply-To: <CAA_Md=t4NR-ocAsnvOmD39XGNAexYVZWsWy4taYbiXU7-N1qEg@mail.gmail.com>
References: <CAA_Md=uAGmnqLkY3g128QcjoBis4VnoTwSUEZpqAws35AwRsHQ@mail.gmail.com>
	<23b1626b3814448b90842a279dc30be5@nysolutions.com>
	<CAA_Md=t4NR-ocAsnvOmD39XGNAexYVZWsWy4taYbiXU7-N1qEg@mail.gmail.com>
Message-ID: <CAEJMVkC+=U50t2nEYrn9CU1OQnXOUcVXKmpzaB8KF=Gbh7T-BQ@mail.gmail.com>

Its just the channel name so don't get tripped up by that, its just named
with some default data about the session.

On Wed, Feb 24, 2016 at 11:33 PM, Brian Chow <freeswitch at digitaldescent.net>
wrote:

> Debug console output: https://pastebin.freeswitch.org/24572
>
> My fqdn only resolves inside the network, but the external-ip is
> configured in FS to use stun, which reports my external ip.
>
> >Is your endpoint registered? How often is it set to reregister?
>
> Yes, and the zoiper default registration expiry is 3600
>
>
>
> >Does your router have a sip alg? Are the ports opened in your firewall?
>
> > ish, no GUI for it, but I did make sure to disable it from the command
> line (Ubiquiti Edgerouter).  No firewall ports are explicitly open, but
> allow bi-directional with existing states.  I do get 2 way audio when
> calling out, and I also can hear the system forwarding my call in to
> voicemail; the voicemail message and prompts.  I know asterisk is different
> from FS, but my asterisk experience has proven that at least with asterisk,
> port forwarding isn't needed.  Also, unfortunately, I cannot get a static
> ip.
>
>
> Thanks!
>
>   -Brian
>
> On Wed, Feb 24, 2016 at 8:31 PM, Moishe Grunstein <max at nysolutions.com>
> wrote:
>
>> It is probably sending call to your extension at domainname, if your
>> external ip is your domain name then you will see the external ip.
>>
>> Is your endpoint registered? How often is it set to reregister?
>>
>> Does your router have a sip alg? Are the ports opened in your firewall?
>>
>> Very hard to guess without a log of a call with debug enabled.
>>
>> sofia global sip trace on
>>
>>
>>
>> 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 *Brian Chow
>> *Sent:* Wednesday, February 24, 2016 10:00 PM
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* [Freeswitch-users] Inbound calls mis-routing routing to
>> internal extension with external IP?
>>
>>
>>
>> Hello all!
>>
>>   I'm new to freeswitch, so I'm sure this is just a newbie configuration
>> error.  Sorry if it's been answered a million times, my google searches
>> always just bring up the standard NAT configuration pages.  I've already
>> followed the confluence page on configuring NAT.
>>
>>
>>
>> My Setup:
>>
>> Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3
>> instance.
>>
>> SIP Provider: Flowroute
>>
>>
>>
>> NAT: extension and freeswitch are on the same network, both of which are
>> behind NAT and connecting to flowroute.  I configured external sip and rtp
>> to use stun entries.  External profile is using $${external_rtp/sip_ip} for
>> ext-rtp/sip respectively.
>>
>>
>>
>> I have one DID configured to go directly to my one extension.  My
>> extension can register just fine.  My extension can dial out.  When I call
>> my cell, the call connects and I have 2 way audio.  When I dial my DID from
>> my cell, I can see the call hitting the FS server, but instead of ringing
>> my extension, it goes straight to my extensions voicemail (which I can just
>> fine).
>>
>>
>>
>> When I look the at the console, it appears (sorry if this is wrong, I'm
>> only a day into free switch) that FS is attempting to route the call to my
>> extension...@ my external ip?
>>
>>
>>
>> I see this line in the console:  2016-02-24 18:56:10.453983 [NOTICE]
>> switch_channel.c:1101 New Channel sofia/internal/<extension>@
>> *<externalip>*:46072
>>
>>
>>
>>
>>
>> Shouldn't that read sofia/internal/<extentsion>@<ip_of_extension>  ?
>>
>>
>>
>> Sofia Status says:
>>
>>
>>
>>                      Name        Type
>>  Data  State
>>
>>
>> =================================================================================================
>>
>>             external-ipv6   profile
>> sip:mod_sofia@[::1]:5080            RUNNING (0)
>>
>>                  external      profile
>> sip:mod_sofia@<external_ip>:5080     RUNNING (0)
>>
>> external::<uuid>          gateway                      sip:<flowrouteid>@
>> sip.flowroute.com          REGED
>>
>>             internal-ipv6    profile
>> sip:mod_sofia@[::1]:5060            RUNNING (0)
>>
>>                  internal       profile
>> sip:mod_sofia@<internal_ip>:5060    RUNNING (0)
>>
>>
>> =================================================================================================
>>
>>
>>
>> If I'm completely off base here, can anyone recommend where I can start
>> looking to change/troubleshoot the issue?  I feel like it's just me missing
>> something, I just can't determine what that might be.
>>
>>
>>
>> Thanks,
>>
>>   -Brian
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

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 freeswitch at digitaldescent.net  Thu Feb 25 18:59:48 2016
From: freeswitch at digitaldescent.net (Brian Chow)
Date: Thu, 25 Feb 2016 07:59:48 -0800
Subject: [Freeswitch-users] Inbound calls mis-routing routing to
 internal extension with external IP?
In-Reply-To: <CAA_Md=v=PGrR_0s90SK_ijZO5eScqJYN+Utqu22nwqbZEQXvLA@mail.gmail.com>
References: <CAA_Md=uAGmnqLkY3g128QcjoBis4VnoTwSUEZpqAws35AwRsHQ@mail.gmail.com>
	<23b1626b3814448b90842a279dc30be5@nysolutions.com>
	<CAA_Md=t4NR-ocAsnvOmD39XGNAexYVZWsWy4taYbiXU7-N1qEg@mail.gmail.com>
	<CAEJMVkC+=U50t2nEYrn9CU1OQnXOUcVXKmpzaB8KF=Gbh7T-BQ@mail.gmail.com>
	<CAA_Md=v=PGrR_0s90SK_ijZO5eScqJYN+Utqu22nwqbZEQXvLA@mail.gmail.com>
Message-ID: <CAA_Md=sTVGHHmAvtV9EV2OvBuLkdnQ3b9wRaSCMYjBwf71p-yA@mail.gmail.com>

Ok, in that case, what do I do next to figure out why inbound calls go
directly to voicemail instead of ringing the extension?
On Feb 25, 2016 7:57 AM, "Brian Chow" <brianchow at digitaldescent.net> wrote:

> Ok, in that case, what do I look at next to see why it goes straight to
> voicemail instead of ringing my extension?
> On Feb 25, 2016 5:52 AM, "Brian West" <brian at freeswitch.org> wrote:
>
>> Its just the channel name so don't get tripped up by that, its just named
>> with some default data about the session.
>>
>> On Wed, Feb 24, 2016 at 11:33 PM, Brian Chow <
>> freeswitch at digitaldescent.net> wrote:
>>
>>> Debug console output: https://pastebin.freeswitch.org/24572
>>>
>>> My fqdn only resolves inside the network, but the external-ip is
>>> configured in FS to use stun, which reports my external ip.
>>>
>>> >Is your endpoint registered? How often is it set to reregister?
>>>
>>> Yes, and the zoiper default registration expiry is 3600
>>>
>>>
>>>
>>> >Does your router have a sip alg? Are the ports opened in your firewall?
>>>
>>> > ish, no GUI for it, but I did make sure to disable it from the command
>>> line (Ubiquiti Edgerouter).  No firewall ports are explicitly open, but
>>> allow bi-directional with existing states.  I do get 2 way audio when
>>> calling out, and I also can hear the system forwarding my call in to
>>> voicemail; the voicemail message and prompts.  I know asterisk is different
>>> from FS, but my asterisk experience has proven that at least with asterisk,
>>> port forwarding isn't needed.  Also, unfortunately, I cannot get a static
>>> ip.
>>>
>>>
>>> Thanks!
>>>
>>>   -Brian
>>>
>>> On Wed, Feb 24, 2016 at 8:31 PM, Moishe Grunstein <max at nysolutions.com>
>>> wrote:
>>>
>>>> It is probably sending call to your extension at domainname, if your
>>>> external ip is your domain name then you will see the external ip.
>>>>
>>>> Is your endpoint registered? How often is it set to reregister?
>>>>
>>>> Does your router have a sip alg? Are the ports opened in your firewall?
>>>>
>>>> Very hard to guess without a log of a call with debug enabled.
>>>>
>>>> sofia global sip trace on
>>>>
>>>>
>>>>
>>>> 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 *Brian
>>>> Chow
>>>> *Sent:* Wednesday, February 24, 2016 10:00 PM
>>>> *To:* freeswitch-users at lists.freeswitch.org
>>>> *Subject:* [Freeswitch-users] Inbound calls mis-routing routing to
>>>> internal extension with external IP?
>>>>
>>>>
>>>>
>>>> Hello all!
>>>>
>>>>   I'm new to freeswitch, so I'm sure this is just a newbie
>>>> configuration error.  Sorry if it's been answered a million times, my
>>>> google searches always just bring up the standard NAT configuration pages.
>>>> I've already followed the confluence page on configuring NAT.
>>>>
>>>>
>>>>
>>>> My Setup:
>>>>
>>>> Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3
>>>> instance.
>>>>
>>>> SIP Provider: Flowroute
>>>>
>>>>
>>>>
>>>> NAT: extension and freeswitch are on the same network, both of which
>>>> are behind NAT and connecting to flowroute.  I configured external sip and
>>>> rtp to use stun entries.  External profile is using $${external_rtp/sip_ip}
>>>> for ext-rtp/sip respectively.
>>>>
>>>>
>>>>
>>>> I have one DID configured to go directly to my one extension.  My
>>>> extension can register just fine.  My extension can dial out.  When I call
>>>> my cell, the call connects and I have 2 way audio.  When I dial my DID from
>>>> my cell, I can see the call hitting the FS server, but instead of ringing
>>>> my extension, it goes straight to my extensions voicemail (which I can just
>>>> fine).
>>>>
>>>>
>>>>
>>>> When I look the at the console, it appears (sorry if this is wrong, I'm
>>>> only a day into free switch) that FS is attempting to route the call to my
>>>> extension...@ my external ip?
>>>>
>>>>
>>>>
>>>> I see this line in the console:  2016-02-24 18:56:10.453983 [NOTICE]
>>>> switch_channel.c:1101 New Channel sofia/internal/<extension>@
>>>> *<externalip>*:46072
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> Shouldn't that read sofia/internal/<extentsion>@<ip_of_extension>  ?
>>>>
>>>>
>>>>
>>>> Sofia Status says:
>>>>
>>>>
>>>>
>>>>                      Name        Type
>>>>    Data  State
>>>>
>>>>
>>>> =================================================================================================
>>>>
>>>>             external-ipv6   profile
>>>> sip:mod_sofia@[::1]:5080            RUNNING (0)
>>>>
>>>>                  external      profile
>>>> sip:mod_sofia@<external_ip>:5080     RUNNING (0)
>>>>
>>>> external::<uuid>          gateway
>>>>  sip:<flowrouteid>@sip.flowroute.com          REGED
>>>>
>>>>             internal-ipv6    profile
>>>> sip:mod_sofia@[::1]:5060            RUNNING (0)
>>>>
>>>>                  internal       profile
>>>> sip:mod_sofia@<internal_ip>:5060    RUNNING (0)
>>>>
>>>>
>>>> =================================================================================================
>>>>
>>>>
>>>>
>>>> If I'm completely off base here, can anyone recommend where I can start
>>>> looking to change/troubleshoot the issue?  I feel like it's just me missing
>>>> something, I just can't determine what that might be.
>>>>
>>>>
>>>>
>>>> Thanks,
>>>>
>>>>   -Brian
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>> 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 max at nysolutions.com  Thu Feb 25 19:12:36 2016
From: max at nysolutions.com (Moishe Grunstein)
Date: Thu, 25 Feb 2016 16:12:36 +0000
Subject: [Freeswitch-users] Inbound calls mis-routing routing to
 internal extension with external IP?
In-Reply-To: <CAA_Md=sTVGHHmAvtV9EV2OvBuLkdnQ3b9wRaSCMYjBwf71p-yA@mail.gmail.com>
References: <CAA_Md=uAGmnqLkY3g128QcjoBis4VnoTwSUEZpqAws35AwRsHQ@mail.gmail.com>
	<23b1626b3814448b90842a279dc30be5@nysolutions.com>
	<CAA_Md=t4NR-ocAsnvOmD39XGNAexYVZWsWy4taYbiXU7-N1qEg@mail.gmail.com>
	<CAEJMVkC+=U50t2nEYrn9CU1OQnXOUcVXKmpzaB8KF=Gbh7T-BQ@mail.gmail.com>
	<CAA_Md=v=PGrR_0s90SK_ijZO5eScqJYN+Utqu22nwqbZEQXvLA@mail.gmail.com>
	<CAA_Md=sTVGHHmAvtV9EV2OvBuLkdnQ3b9wRaSCMYjBwf71p-yA@mail.gmail.com>
Message-ID: <bf0ede9ef70640818f0607ceb46f5434@nysolutions.com>

Does internal call also go direct to voicemail?
Are you sure you don?t have a *99 before the extension number?

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 Brian Chow
Sent: Thursday, February 25, 2016 11:00 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP?


Ok, in that case, what do I do next to figure out why inbound calls go directly to voicemail instead of ringing the extension?
On Feb 25, 2016 7:57 AM, "Brian Chow" <brianchow at digitaldescent.net<mailto:brianchow at digitaldescent.net>> wrote:

Ok, in that case, what do I look at next to see why it goes straight to voicemail instead of ringing my extension?
On Feb 25, 2016 5:52 AM, "Brian West" <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:
Its just the channel name so don't get tripped up by that, its just named with some default data about the session.

On Wed, Feb 24, 2016 at 11:33 PM, Brian Chow <freeswitch at digitaldescent.net<mailto:freeswitch at digitaldescent.net>> wrote:
Debug console output: https://pastebin.freeswitch.org/24572

My fqdn only resolves inside the network, but the external-ip is configured in FS to use stun, which reports my external ip.

>Is your endpoint registered? How often is it set to reregister?
Yes, and the zoiper default registration expiry is 3600

>Does your router have a sip alg? Are the ports opened in your firewall?
> ish, no GUI for it, but I did make sure to disable it from the command line (Ubiquiti Edgerouter).  No firewall ports are explicitly open, but allow bi-directional with existing states.  I do get 2 way audio when calling out, and I also can hear the system forwarding my call in to voicemail; the voicemail message and prompts.  I know asterisk is different from FS, but my asterisk experience has proven that at least with asterisk, port forwarding isn't needed.  Also, unfortunately, I cannot get a static ip.

Thanks!
  -Brian

On Wed, Feb 24, 2016 at 8:31 PM, Moishe Grunstein <max at nysolutions.com<mailto:max at nysolutions.com>> wrote:
It is probably sending call to your extension at domainname, if your external ip is your domain name then you will see the external ip.
Is your endpoint registered? How often is it set to reregister?
Does your router have a sip alg? Are the ports opened in your firewall?
Very hard to guess without a log of a call with debug enabled.
sofia global sip trace on

Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650<tel:212.400.7650> 888.IPPBX.US<http://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> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Brian Chow
Sent: Wednesday, February 24, 2016 10:00 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP?

Hello all!
  I'm new to freeswitch, so I'm sure this is just a newbie configuration error.  Sorry if it's been answered a million times, my google searches always just bring up the standard NAT configuration pages.  I've already followed the confluence page on configuring NAT.

My Setup:
Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3 instance.
SIP Provider: Flowroute

NAT: extension and freeswitch are on the same network, both of which are behind NAT and connecting to flowroute.  I configured external sip and rtp to use stun entries.  External profile is using $${external_rtp/sip_ip} for ext-rtp/sip respectively.

I have one DID configured to go directly to my one extension.  My extension can register just fine.  My extension can dial out.  When I call my cell, the call connects and I have 2 way audio.  When I dial my DID from my cell, I can see the call hitting the FS server, but instead of ringing my extension, it goes straight to my extensions voicemail (which I can just fine).

When I look the at the console, it appears (sorry if this is wrong, I'm only a day into free switch) that FS is attempting to route the call to my extension...@ my external ip?

I see this line in the console:  2016-02-24 18:56:10.453983 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/<extension>@<externalip>:46072


Shouldn't that read sofia/internal/<extentsion>@<ip_of_extension>  ?

Sofia Status says:

                     Name        Type                                      Data  State
=================================================================================================
            external-ipv6   profile                   sip:mod_sofia@[::1]:5080            RUNNING (0)
                 external      profile           sip:mod_sofia@<external_ip>:5080<sip:mod_sofia@%3cexternal_ip%3e:5080>     RUNNING (0)
external::<uuid>          gateway                      sip:<flowrouteid>@sip.flowroute.com<http://sip.flowroute.com>          REGED
            internal-ipv6    profile                   sip:mod_sofia@[::1]:5060            RUNNING (0)
                 internal       profile            sip:mod_sofia@<internal_ip>:5060<sip:mod_sofia@%3cinternal_ip%3e:5060>    RUNNING (0)
=================================================================================================

If I'm completely off base here, can anyone recommend where I can start looking to change/troubleshoot the issue?  I feel like it's just me missing something, I just can't determine what that might be.

Thanks,
  -Brian

_________________________________________________________________________
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
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--

Brian West
brian at freeswitch.org<mailto:brian at freeswitch.org>

[http://billing.freeswitch.org/templates/default/img/whmcslogo.png]

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

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

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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
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Official FreeSWITCH Sites
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From aqsyounas at gmail.com  Thu Feb 25 19:37:26 2016
From: aqsyounas at gmail.com (Aqs Younas)
Date: Thu, 25 Feb 2016 21:37:26 +0500
Subject: [Freeswitch-users] Unable to detect beep with mod_avmd
In-Reply-To: <CAKzWOxXVV5fjyOEP=3mDbNH29Rv2=7iesPbdPZkzypriPVcWzg@mail.gmail.com>
References: <CAGVOjVUvrPVbO_jh97f0gKcubbOonvDzpo-7=e3_0i3frSBZfA@mail.gmail.com>
	<CAKzWOxXVV5fjyOEP=3mDbNH29Rv2=7iesPbdPZkzypriPVcWzg@mail.gmail.com>
Message-ID: <CAGVOjVXxD=D8--s-1gz44Vi-Aa3d4dNcVv=ULBkyiOhP-xXwGg@mail.gmail.com>

Thank You Much. I was testing it on single leg call. Starting avmd on
outbound leg(of two leg call) perfectly detected the beep.


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On 24 February 2016 at 06:27, Michael Collins <msc at freeswitch.org> wrote:

> It definitely works, I just don't think it works on a one-legged call.
> From fs_cli try this:
>
> originate loopback/1001 &playback(silence_stream://60000)
>
> You'll have two uuids. Try the avmd app on each one. It will detect it on
> the "outbound" leg but not the "inbound" leg.
>
> -MSC
>
> On Tue, Feb 23, 2016 at 5:34 AM, Aqs Younas <aqsyounas at gmail.com> wrote:
>
>> Hi, I am using avmd to detect beep over a channel. Beep is generated by
>> below script.
>>
>> <extension name="10014">
>>        <condition field="destination_number" expression="^1001$">
>>         <action application="playback"
>> data="tone_stream://L=5;%(500,6850,850)"/>
>>          <action application="hangup"/>
>>        </condition>
>>   </extension>
>>
>> and starts avmd on fs_cli as.
>>
>> freeswitch at debian>avmd 30d99731-4ff3-4f02-ad4b-85a3a8b9d600 start
>>
>>
>> I could proper hear beeps over the phone but does not see avmd detecting
>> this. Does not see anything like "beep detected" in logs.
>>
>> I am made several attempts but unable to detect beep even a single time.
>>
>> Is there something wrong that i am doing?
>>
>> Thanks in advance.
>>
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 joelists at tm.net.uk  Thu Feb 25 20:00:08 2016
From: joelists at tm.net.uk (Joseph Waite)
Date: Thu, 25 Feb 2016 17:00:08 +0000
Subject: [Freeswitch-users] Freeswitch/FusionPBX sales force integration
Message-ID: <90B18E57-F6AF-405F-A65B-A034F2AAE069@tm.net.uk>

Evening all

Does anyone know of an integration with sales force?

Looking for click to dial and screen pops primarily. With some kind of call recording integration being ideal.

Open source preferred, commercial solutions considered!

Joe Waite

From aqsyounas at gmail.com  Thu Feb 25 20:13:48 2016
From: aqsyounas at gmail.com (Aqs Younas)
Date: Thu, 25 Feb 2016 22:13:48 +0500
Subject: [Freeswitch-users] How to kill call/channel on the reception of
	CUSTOM avmd::beep
Message-ID: <CAGVOjVUT0L8uZ_Fspv9CAnq1Vnn1tXyMKnLmmfSYW5qDD+-zKQ@mail.gmail.com>

Hi,

I have subscribed to CUSTOM avmd::beep and this is what I get.

        "Event-Subclass":       "avmd::beep",
        "Event-Name":   "CUSTOM",
        "Core-UUID":    "11e64e66-f81b-4d23-a8e1-e6af0eaa0bbc",
        "FreeSWITCH-Hostname":  "debian",
        "FreeSWITCH-Switchname":        "debian",
        "FreeSWITCH-IPv4":      "192.168.10.59",
        "FreeSWITCH-IPv6":      "::1",
        "Event-Date-Local":     "2016-02-25 10:46:17",
        "Event-Date-GMT":       "Thu, 25 Feb 2016 15:46:17 GMT",
        "Event-Date-Timestamp": "1456415177487145",
        "Event-Calling-File":   "mod_avmd.c",
        "Event-Calling-Function":       "avmd_process",
        "Event-Calling-Line-Number":    "550",
        "Event-Sequence":       "3174",
        "Beep-Status":  "stop",
        "Unique-ID":    "80d33754-b234-47e2-b7c5-16ffa20f359c",
        "call-command": "avmd"

Now I need to hangup that call(beep detected), but could not find any event
variable connected to the call.

How can i kill the call.

Any pointer is much appreciated.

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From max at nysolutions.com  Thu Feb 25 20:26:08 2016
From: max at nysolutions.com (Moishe Grunstein)
Date: Thu, 25 Feb 2016 17:26:08 +0000
Subject: [Freeswitch-users] Freeswitch/FusionPBX sales force integration
In-Reply-To: <90B18E57-F6AF-405F-A65B-A034F2AAE069@tm.net.uk>
References: <90B18E57-F6AF-405F-A65B-A034F2AAE069@tm.net.uk>
Message-ID: <8e62b7d38f7e401eb73f10ad61120152@nysolutions.com>

There are several solutions that work with any system, they talk to the endpoint. You can look at http://camrivox.com/products/flexor-cti-salesforce/
Not opensource


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 Joseph Waite
Sent: Thursday, February 25, 2016 12:00 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Freeswitch/FusionPBX sales force integration

Evening all

Does anyone know of an integration with sales force?

Looking for click to dial and screen pops primarily. With some kind of call recording integration being ideal.

Open source preferred, commercial solutions considered!

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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at 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 mario_fs at mgtech.com  Thu Feb 25 21:16:15 2016
From: mario_fs at mgtech.com (Mario G)
Date: Thu, 25 Feb 2016 10:16:15 -0800
Subject: [Freeswitch-users] major build change in master.
In-Reply-To: <F7862F99-22B2-4165-8EE3-9872E848B14E@jerris.com>
References: <F7862F99-22B2-4165-8EE3-9872E848B14E@jerris.com>
Message-ID: <8C3827C6-22F2-45BB-BFAA-6AB64BD67E12@mgtech.com>

This fixed OS X VPX issue, aok now. Questions:

I assume this means YUV and VPX can be removed as prerequisites?

Will this be one for 1.6 as well?

Just asking so I can update the wiki and Applescript installer as needed. Thanks!


> On Feb 24, 2016, at 3:49 PM, Michael Jerris <mike at jerris.com> wrote:
> 
> please note this important change that went into master today.  As of today, we are no longer using system versions of libyuv and libvpx due to major conflicts with system versions of these libraries.  These are now built static into the freeswitch core.  Also note, mod_vpx no longer exists, it is automatically loaded as part of the core and you will no longer have mod_vpx.so or have to manually load it.  I'll have more details coming, but let me know if you have any questions.  Please note for anyone doing cross compiling, this probably needs a bit more work for you, I'll be working on fixing that this week.
> 
> Mike
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 bote_radio at botecomm.com  Thu Feb 25 21:26:09 2016
From: bote_radio at botecomm.com (Bote Man)
Date: Thu, 25 Feb 2016 13:26:09 -0500
Subject: [Freeswitch-users] Getting 503 when starting automatically
	via	systemd
In-Reply-To: <BN3PR13MB0785EEACC5A82AAF57D9B2F1D7A60@BN3PR13MB0785.namprd13.prod.outlook.com>
References: <BN3PR13MB0785EEACC5A82AAF57D9B2F1D7A60@BN3PR13MB0785.namprd13.prod.outlook.com>
Message-ID: <01de01d16ff9$ffe124e0$ffa36ea0$@botecomm.com>

A 503 error, huh? Weird. Check for permissions problems.

My guess is that you are building as "root" so when you first ran it
manually all the databases and log files were created with root as owner.
But when it boots systemd is starting it as root and then dropping
privileges to the freeswitch user as it should. After dropping privileges it
can't access its config files or open ports necessary to function.

Again, this is just my S.W.A.G. but it has happened enough times in the past
that it's a pretty good guess. If it bears out you might want to check the
instructions on this deprecated page:
https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video#Fr
eeSWITCH1.6Video-StartingFreeSWITCH

Also, once you achieve success there are MANY other RPi users out there who
would love to see your recipe for success.
https://freeswitch.org/confluence/display/FREESWITCH/Raspberry+Pi

Hope this helps.

Bote


> -----Original Message-----
> From:  freeswitch.opencode at spamgourmet.com
> Sent: Thursday, 25 February, 2016 02:36
> Subject: [Freeswitch-users] Getting 503 when starting automatically via
systemd
> 
> I'm using Arch Linux (specifically Arch Linux ARM on a Raspberry Pi Zero).
I'm
> successfully done a make and install for freeswitch 1.6.6 and when I run
it
> manually (freeswitch -nonat), it works fine. Also, when I start it
manually via
> systemd (systemctl start freeswitch), it still works fine.
> 
> However, when I enable freeswitch.service to start automatically and
> reboot, the freeswitch log shows 503 errors trying to register out to my
> gateway, and my internal client (an Obi200) gets a 503 when trying to
register
> in to FreeSWITCH as well; this keeps happening repeatedly (waiting 30
> seconds and then failing again). However, if I restart the service
(systemctl
> restart freeswitch), connectivity then starts working.
> 
> Would the 503 be what's reported up if there's no network connectivity
> either way?
> 
> I'm using the Debian systemd *freeswitch.service file (with very minor
> tweaks for my configuration on Arch Linux). Has anyone encountered this
> problem before (on Debian, Arch Linux, or elsewhere)? The overall .service
> file is very similar to sshd.service, which doesn't have connection
problems,
> so my guess is there's something specific to FreeSWITCH going on. If this
is
> Arch Linux-specific and I ask in their forums, is there anything different
about
> how FreeSWITCH handles connectivity (perhaps caching some kind of
> handle) that would help point things in the right direction?
> 
> Thanks,
> 
> David
> 



From bote_radio at botecomm.com  Thu Feb 25 21:39:27 2016
From: bote_radio at botecomm.com (Bote Man)
Date: Thu, 25 Feb 2016 13:39:27 -0500
Subject: [Freeswitch-users] major build change in master.
In-Reply-To: <8C3827C6-22F2-45BB-BFAA-6AB64BD67E12@mgtech.com>
References: <F7862F99-22B2-4165-8EE3-9872E848B14E@jerris.com>
	<8C3827C6-22F2-45BB-BFAA-6AB64BD67E12@mgtech.com>
Message-ID: <01e201d16ffb$db648f10$922dad30$@botecomm.com>

Mike, I put a note to this effect on the Debian 8 installation page.

I'm glad I caught this as I have been on a customer's site a lot and have
not been able to keep with the -users mailing list. Lemme know if this
changes so I can update it.

Thanks.

Bote


> -----Original Message-----
> From: Mario G
> Sent: Thursday, 25 February, 2016 13:16
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] major build change in master.
> 
> This fixed OS X VPX issue, aok now. Questions:
> 
> I assume this means YUV and VPX can be removed as prerequisites?
> 
> Will this be one for 1.6 as well?
> 
> Just asking so I can update the wiki and Applescript installer as needed.
> Thanks!
> 
> 
> > On Feb 24, 2016, at 3:49 PM, Michael Jerris <mike at jerris.com> wrote:
> >
> > please note this important change that went into master today.  As of
> today, we are no longer using system versions of libyuv and libvpx due to
> major conflicts with system versions of these libraries.  These are now
built
> static into the freeswitch core.  Also note, mod_vpx no longer exists, it
is
> automatically loaded as part of the core and you will no longer have
> mod_vpx.so or have to manually load it.  I'll have more details coming,
but let
> me know if you have any questions.  Please note for anyone doing cross
> compiling, this probably needs a bit more work for you, I'll be working on
> fixing that this week.
> >
> > Mike
> >



From blasterjr at gmail.com  Thu Feb 25 21:43:10 2016
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Thu, 25 Feb 2016 11:43:10 -0700
Subject: [Freeswitch-users] How to kill call/channel on the reception of
 CUSTOM avmd::beep
In-Reply-To: <CAGVOjVUT0L8uZ_Fspv9CAnq1Vnn1tXyMKnLmmfSYW5qDD+-zKQ@mail.gmail.com>
References: <CAGVOjVUT0L8uZ_Fspv9CAnq1Vnn1tXyMKnLmmfSYW5qDD+-zKQ@mail.gmail.com>
Message-ID: <CAO3hyzG_xaLhUHT-qEgB2nxQYsAWjNO0QCGdWdA=Kfxf0zWL3g@mail.gmail.com>

uuid_hangup the value of the Unique-ID field possibly?

On Thu, Feb 25, 2016 at 10:13 AM, Aqs Younas <aqsyounas at gmail.com> wrote:

> Hi,
>
> I have subscribed to CUSTOM avmd::beep and this is what I get.
>
>         "Event-Subclass":       "avmd::beep",
>         "Event-Name":   "CUSTOM",
>         "Core-UUID":    "11e64e66-f81b-4d23-a8e1-e6af0eaa0bbc",
>         "FreeSWITCH-Hostname":  "debian",
>         "FreeSWITCH-Switchname":        "debian",
>         "FreeSWITCH-IPv4":      "192.168.10.59",
>         "FreeSWITCH-IPv6":      "::1",
>         "Event-Date-Local":     "2016-02-25 10:46:17",
>         "Event-Date-GMT":       "Thu, 25 Feb 2016 15:46:17 GMT",
>         "Event-Date-Timestamp": "1456415177487145",
>         "Event-Calling-File":   "mod_avmd.c",
>         "Event-Calling-Function":       "avmd_process",
>         "Event-Calling-Line-Number":    "550",
>         "Event-Sequence":       "3174",
>         "Beep-Status":  "stop",
>         "Unique-ID":    "80d33754-b234-47e2-b7c5-16ffa20f359c",
>         "call-command": "avmd"
>
> Now I need to hangup that call(beep detected), but could not find any
> event variable connected to the call.
>
> How can i kill the call.
>
> Any pointer is much appreciated.
>
> This email has been sent from a virus-free computer protected by Avast.
> www.avast.com
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
> <#-1458571440_DDB4FAA8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Feb 25 21:48:27 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 25 Feb 2016 13:48:27 -0500
Subject: [Freeswitch-users] major build change in master.
In-Reply-To: <8C3827C6-22F2-45BB-BFAA-6AB64BD67E12@mgtech.com>
References: <F7862F99-22B2-4165-8EE3-9872E848B14E@jerris.com>
	<8C3827C6-22F2-45BB-BFAA-6AB64BD67E12@mgtech.com>
Message-ID: <2C14558E-FBD9-4532-A1FC-2272DF36DB71@jerris.com>

This will remove the prereqs for all new versions released after this date, so the newest 1.6 for example will not need those anymore, all previous requirements for previous versions stay the same.

> On Feb 25, 2016, at 1:16 PM, Mario G <mario_fs at mgtech.com> wrote:
> 
> This fixed OS X VPX issue, aok now. Questions:
> 
> I assume this means YUV and VPX can be removed as prerequisites?
> 
> Will this be one for 1.6 as well?
> 
> Just asking so I can update the wiki and Applescript installer as needed. Thanks!
> 
> 
>> On Feb 24, 2016, at 3:49 PM, Michael Jerris <mike at jerris.com> wrote:
>> 
>> please note this important change that went into master today.  As of today, we are no longer using system versions of libyuv and libvpx due to major conflicts with system versions of these libraries.  These are now built static into the freeswitch core.  Also note, mod_vpx no longer exists, it is automatically loaded as part of the core and you will no longer have mod_vpx.so or have to manually load it.  I'll have more details coming, but let me know if you have any questions.  Please note for anyone doing cross compiling, this probably needs a bit more work for you, I'll be working on fixing that this week.
>> 
>> Mike


From mike at jerris.com  Thu Feb 25 21:49:33 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 25 Feb 2016 13:49:33 -0500
Subject: [Freeswitch-users] major build change in master.
In-Reply-To: <01e201d16ffb$db648f10$922dad30$@botecomm.com>
References: <F7862F99-22B2-4165-8EE3-9872E848B14E@jerris.com>
	<8C3827C6-22F2-45BB-BFAA-6AB64BD67E12@mgtech.com>
	<01e201d16ffb$db648f10$922dad30$@botecomm.com>
Message-ID: <8D88662E-BB53-49F2-B003-2C9288A3FD03@jerris.com>

we are working on how exactly this will effect packaging and install instructions, but we will no longer have yuv or vpx pre-reqs and the requirement to rebuild libav against our vpx is also removed.

> On Feb 25, 2016, at 1:39 PM, Bote Man <bote_radio at botecomm.com> wrote:
> 
> Mike, I put a note to this effect on the Debian 8 installation page.
> 
> I'm glad I caught this as I have been on a customer's site a lot and have
> not been able to keep with the -users mailing list. Lemme know if this
> changes so I can update it.
> 
> Thanks.
> 
> Bote
> 
> 
>> -----Original Message-----
>> From: Mario G
>> Sent: Thursday, 25 February, 2016 13:16
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] major build change in master.
>> 
>> This fixed OS X VPX issue, aok now. Questions:
>> 
>> I assume this means YUV and VPX can be removed as prerequisites?
>> 
>> Will this be one for 1.6 as well?
>> 
>> Just asking so I can update the wiki and Applescript installer as needed.
>> Thanks!
>> 
>> 
>>> On Feb 24, 2016, at 3:49 PM, Michael Jerris <mike at jerris.com> wrote:
>>> 
>>> please note this important change that went into master today.  As of
>> today, we are no longer using system versions of libyuv and libvpx due to
>> major conflicts with system versions of these libraries.  These are now
> built
>> static into the freeswitch core.  Also note, mod_vpx no longer exists, it
> is
>> automatically loaded as part of the core and you will no longer have
>> mod_vpx.so or have to manually load it.  I'll have more details coming,
> but let
>> me know if you have any questions.  Please note for anyone doing cross
>> compiling, this probably needs a bit more work for you, I'll be working on
>> fixing that this week.
>>> 
>>> Mike
>>> 



From aqsyounas at gmail.com  Thu Feb 25 23:01:12 2016
From: aqsyounas at gmail.com (Aqs Younas)
Date: Fri, 26 Feb 2016 01:01:12 +0500
Subject: [Freeswitch-users] How to kill call/channel on the reception of
 CUSTOM avmd::beep
In-Reply-To: <CAO3hyzG_xaLhUHT-qEgB2nxQYsAWjNO0QCGdWdA=Kfxf0zWL3g@mail.gmail.com>
References: <CAGVOjVUT0L8uZ_Fspv9CAnq1Vnn1tXyMKnLmmfSYW5qDD+-zKQ@mail.gmail.com>
	<CAO3hyzG_xaLhUHT-qEgB2nxQYsAWjNO0QCGdWdA=Kfxf0zWL3g@mail.gmail.com>
Message-ID: <CAGVOjVXa723vJc+hq-fYzA0X0yvF7im+Bu4u18DJHSa-YS4ByA@mail.gmail.com>

Thanks for your Reply.

freeswitch at debian> uuid_hangup Unique-ID
Unknown Command: uuid_hangup 11e64e66-f81b-4d23-a8e1-e6af0eaa0bbc

Tried to kill.

freeswitch at debian> uuid_kill Unique-ID

-ERR No such channel!

Any suggestion?

On 25 February 2016 at 23:43, Chris Tunbridge <blasterjr at gmail.com> wrote:

> uuid_hangup the value of the Unique-ID field possibly?
>
> On Thu, Feb 25, 2016 at 10:13 AM, Aqs Younas <aqsyounas at gmail.com> wrote:
>
>> Hi,
>>
>> I have subscribed to CUSTOM avmd::beep and this is what I get.
>>
>>         "Event-Subclass":       "avmd::beep",
>>         "Event-Name":   "CUSTOM",
>>         "Core-UUID":    "11e64e66-f81b-4d23-a8e1-e6af0eaa0bbc",
>>         "FreeSWITCH-Hostname":  "debian",
>>         "FreeSWITCH-Switchname":        "debian",
>>         "FreeSWITCH-IPv4":      "192.168.10.59",
>>         "FreeSWITCH-IPv6":      "::1",
>>         "Event-Date-Local":     "2016-02-25 10:46:17",
>>         "Event-Date-GMT":       "Thu, 25 Feb 2016 15:46:17 GMT",
>>         "Event-Date-Timestamp": "1456415177487145",
>>         "Event-Calling-File":   "mod_avmd.c",
>>         "Event-Calling-Function":       "avmd_process",
>>         "Event-Calling-Line-Number":    "550",
>>         "Event-Sequence":       "3174",
>>         "Beep-Status":  "stop",
>>         "Unique-ID":    "80d33754-b234-47e2-b7c5-16ffa20f359c",
>>         "call-command": "avmd"
>>
>> Now I need to hangup that call(beep detected), but could not find any
>> event variable connected to the call.
>>
>> How can i kill the call.
>>
>> Any pointer is much appreciated.
>>
>> This email has been sent from a virus-free computer protected by Avast.
>> www.avast.com
>> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
>> <#-2019354057_-1458571440_DDB4FAA8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Thu Feb 25 23:58:53 2016
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Fri, 26 Feb 2016 04:58:53 +0800
Subject: [Freeswitch-users] gsmopen problem: INCOMPATIBLE_DESTINATION
In-Reply-To: <CABMX_w-nfvq2aeF8VhXZmQGi-PjLURnXtrHT55_XAV32XVCzrQ@mail.gmail.com>
References: <CAHKqMb7MosEn_kA5KYhME+rfG5G+3Yqwiuoexanww=CFZb+N8g@mail.gmail.com>
	<CABMX_w-nfvq2aeF8VhXZmQGi-PjLURnXtrHT55_XAV32XVCzrQ@mail.gmail.com>
Message-ID: <CAHKqMb6UrRJ8+wZcg650sNiixsb0XBCVzdZwFffB5XDdUhxwOg@mail.gmail.com>

I installed version 1.7.  Somehow I got a tip to use absolute_codec_string
in my dialplan.  "set"ting failed.  Realizing that non-L16 codecs has to be
in B-leg,  "export"ing absolute_codec_string worked.  I'll try your
suggestion because the variable is new to me.  Thanks!

====================================================
MagicBox.PH <http://magicboxph.x10.bz> -
*keeps Filipino families in touch worldwide for free*Lapulapu City, Phils
Phone: +63-32-3401807,
Mobile: +63-920-6373450
*USA DID# :* (646)547-1225
*Worldwide:* [**011*][*63-32-3401807#*] via any 200+ Access Numbers (37
Countries) <http://www.snipurl.com/magicbox-access>


On Thu, Feb 25, 2016 at 7:55 PM, ?talo Rossi <italo at freeswitch.org> wrote:

> What version are you using? Try setting
> media_mix_inbound_outbound_codecs=true
>
> On Tue, Feb 23, 2016 at 12:23 AM, Nandy Dagondon <nandy1925 at gmail.com>
> wrote:
>
>> Hello guys,
>>
>> I installed mod_gsmopen. When testing incoming call, it hangs up with
>> this error message INCOMPATIBLE_DESTINATION.  I understand this is a codec
>> negotiation issue.
>>
>> Looking at the CDR - the gsmopen presented L16 8000 in the SDP.  By
>> default, FS does early negotiation.
>>
>> Snippet of my internal sip_profile:
>> <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="presence-privacy" value="$${presence_privacy}"/>
>> <param name="inbound-codec-negotiation" value="generous"/>
>>
>> Take note: there is no disable-transcoding parameter which I understand
>> FS is transcodind.
>>
>> In vars.xml:
>>
>> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h
>> ,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
>> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>>
>> I tried to add L16 at the end of global_codec_prefs which is unnecessary,
>> isn't it? But it didn't work, too.
>>
>> Finally, I added L16 in the softphone, it worked.
>>
>> I have installed gsmopen before and it worked (transcoding) with IP
>> phones.
>>
>> Is there anything I may have missed?  I appreciate for any leads.
>>
>> Thank you,
>>
>> /Nandy
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 blasterjr at gmail.com  Fri Feb 26 00:10:54 2016
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Thu, 25 Feb 2016 14:10:54 -0700
Subject: [Freeswitch-users] How to kill call/channel on the reception of
 CUSTOM avmd::beep
In-Reply-To: <CAGVOjVXa723vJc+hq-fYzA0X0yvF7im+Bu4u18DJHSa-YS4ByA@mail.gmail.com>
References: <CAGVOjVUT0L8uZ_Fspv9CAnq1Vnn1tXyMKnLmmfSYW5qDD+-zKQ@mail.gmail.com>
	<CAO3hyzG_xaLhUHT-qEgB2nxQYsAWjNO0QCGdWdA=Kfxf0zWL3g@mail.gmail.com>
	<CAGVOjVXa723vJc+hq-fYzA0X0yvF7im+Bu4u18DJHSa-YS4ByA@mail.gmail.com>
Message-ID: <CAO3hyzHdi2wR-Ehtr4no_PoBOf0MHNA6H8bmb6NH47r7=zJt6g@mail.gmail.com>

My guess is the UUID of that call is not the same as the Unique-ID, i don't
think it'd be Core-UUID either.  I haven't personally played with mod_amvd
or the events from it.

On Thu, Feb 25, 2016 at 1:01 PM, Aqs Younas <aqsyounas at gmail.com> wrote:

> Thanks for your Reply.
>
> freeswitch at debian> uuid_hangup Unique-ID
> Unknown Command: uuid_hangup 11e64e66-f81b-4d23-a8e1-e6af0eaa0bbc
>
> Tried to kill.
>
> freeswitch at debian> uuid_kill Unique-ID
>
> -ERR No such channel!
>
> Any suggestion?
>
> On 25 February 2016 at 23:43, Chris Tunbridge <blasterjr at gmail.com> wrote:
>
>> uuid_hangup the value of the Unique-ID field possibly?
>>
>> On Thu, Feb 25, 2016 at 10:13 AM, Aqs Younas <aqsyounas at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> I have subscribed to CUSTOM avmd::beep and this is what I get.
>>>
>>>         "Event-Subclass":       "avmd::beep",
>>>         "Event-Name":   "CUSTOM",
>>>         "Core-UUID":    "11e64e66-f81b-4d23-a8e1-e6af0eaa0bbc",
>>>         "FreeSWITCH-Hostname":  "debian",
>>>         "FreeSWITCH-Switchname":        "debian",
>>>         "FreeSWITCH-IPv4":      "192.168.10.59",
>>>         "FreeSWITCH-IPv6":      "::1",
>>>         "Event-Date-Local":     "2016-02-25 10:46:17",
>>>         "Event-Date-GMT":       "Thu, 25 Feb 2016 15:46:17 GMT",
>>>         "Event-Date-Timestamp": "1456415177487145",
>>>         "Event-Calling-File":   "mod_avmd.c",
>>>         "Event-Calling-Function":       "avmd_process",
>>>         "Event-Calling-Line-Number":    "550",
>>>         "Event-Sequence":       "3174",
>>>         "Beep-Status":  "stop",
>>>         "Unique-ID":    "80d33754-b234-47e2-b7c5-16ffa20f359c",
>>>         "call-command": "avmd"
>>>
>>> Now I need to hangup that call(beep detected), but could not find any
>>> event variable connected to the call.
>>>
>>> How can i kill the call.
>>>
>>> Any pointer is much appreciated.
>>>
>>> This email has been sent from a virus-free computer protected by Avast.
>>> www.avast.com
>>> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
>>> <#-1363453803_-2019354057_-1458571440_DDB4FAA8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Fri Feb 26 00:15:05 2016
From: aqsyounas at gmail.com (Aqs Younas)
Date: Fri, 26 Feb 2016 02:15:05 +0500
Subject: [Freeswitch-users] How to kill call/channel on the reception of
 CUSTOM avmd::beep
In-Reply-To: <CAGVOjVXa723vJc+hq-fYzA0X0yvF7im+Bu4u18DJHSa-YS4ByA@mail.gmail.com>
References: <CAGVOjVUT0L8uZ_Fspv9CAnq1Vnn1tXyMKnLmmfSYW5qDD+-zKQ@mail.gmail.com>
	<CAO3hyzG_xaLhUHT-qEgB2nxQYsAWjNO0QCGdWdA=Kfxf0zWL3g@mail.gmail.com>
	<CAGVOjVXa723vJc+hq-fYzA0X0yvF7im+Bu4u18DJHSa-YS4ByA@mail.gmail.com>
Message-ID: <CAGVOjVUzZQ2sVOejJyErMNJTRFOMrzeb+v+1gasDZNM7DUFvWQ@mail.gmail.com>

My fault. Got it working with uuid_kill Unique-ID

Thanks.

On 26 February 2016 at 01:01, Aqs Younas <aqsyounas at gmail.com> wrote:

> Thanks for your Reply.
>
> freeswitch at debian> uuid_hangup Unique-ID
> Unknown Command: uuid_hangup 11e64e66-f81b-4d23-a8e1-e6af0eaa0bbc
>
> Tried to kill.
>
> freeswitch at debian> uuid_kill Unique-ID
>
> -ERR No such channel!
>
> Any suggestion?
>
> On 25 February 2016 at 23:43, Chris Tunbridge <blasterjr at gmail.com> wrote:
>
>> uuid_hangup the value of the Unique-ID field possibly?
>>
>> On Thu, Feb 25, 2016 at 10:13 AM, Aqs Younas <aqsyounas at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> I have subscribed to CUSTOM avmd::beep and this is what I get.
>>>
>>>         "Event-Subclass":       "avmd::beep",
>>>         "Event-Name":   "CUSTOM",
>>>         "Core-UUID":    "11e64e66-f81b-4d23-a8e1-e6af0eaa0bbc",
>>>         "FreeSWITCH-Hostname":  "debian",
>>>         "FreeSWITCH-Switchname":        "debian",
>>>         "FreeSWITCH-IPv4":      "192.168.10.59",
>>>         "FreeSWITCH-IPv6":      "::1",
>>>         "Event-Date-Local":     "2016-02-25 10:46:17",
>>>         "Event-Date-GMT":       "Thu, 25 Feb 2016 15:46:17 GMT",
>>>         "Event-Date-Timestamp": "1456415177487145",
>>>         "Event-Calling-File":   "mod_avmd.c",
>>>         "Event-Calling-Function":       "avmd_process",
>>>         "Event-Calling-Line-Number":    "550",
>>>         "Event-Sequence":       "3174",
>>>         "Beep-Status":  "stop",
>>>         "Unique-ID":    "80d33754-b234-47e2-b7c5-16ffa20f359c",
>>>         "call-command": "avmd"
>>>
>>> Now I need to hangup that call(beep detected), but could not find any
>>> event variable connected to the call.
>>>
>>> How can i kill the call.
>>>
>>> Any pointer is much appreciated.
>>>
>>> This email has been sent from a virus-free computer protected by Avast.
>>> www.avast.com
>>> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
>>> <#996209302_-2019354057_-1458571440_DDB4FAA8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 freeswitch.opencode at spamgourmet.com  Thu Feb 25 22:01:41 2016
From: freeswitch.opencode at spamgourmet.com (freeswitch.opencode at spamgourmet.com)
Date: Thu, 25 Feb 2016 19:01:41 +0000
Subject: [Freeswitch-users]  Getting 503 when starting automatically
Message-ID: <BN3PR13MB078550D45CAAEDE31D682A60D7A60@BN3PR13MB0785.namprd13.prod.outlook.com>

On Thu, 2016-02-25 at 21:26:09 +0300, Bote wrote:
> A 503 error, huh? Weird. Check for permissions problems.

> My guess is that you are building as "root" so when you first ran it
> manually all the databases and log files were created with root as owner.
> But when it boots systemd is starting it as root and then dropping
> privileges to the freeswitch user as it should. After dropping privileges it
> can't access its config files or open ports necessary to function.

Thanks for the suggestion. I compiled as a normal user using Arch Linux's makepkg tool and installed using its pacman tool, so I don't think that's the problem. I'm also letting system start freeswitch.service with its default user setting (as root).

The weirdest thing is that it does start and connect fine using systemd when make a request after boot to have systemd start it (systemctl start freeswitch or systemctl restart freeswitch). In both cases, systemd is controlling process identity, so I don't think it's a permissions issue. My best guess is there's some kind of race condition at boot and if FreeSWITCH starts too early, it gets a cached but invalid handle of some sort that doesn't allow communicating in or out. This is just a guess though.

Thanks,

David


From lists at kavun.ch  Fri Feb 26 02:13:42 2016
From: lists at kavun.ch (Emrah)
Date: Fri, 26 Feb 2016 00:13:42 +0100
Subject: [Freeswitch-users] SRTP breaks my TLS session
Message-ID: <84F64031-B943-4671-A0BC-66FC02DF7C51@kavun.ch>

Hello list,
I thought I had solved this issue by reducing my codec list  to a minimum, but it still persists, unfortunately. This was to reduce the TLS packet size.
Anytime I enable SRTP on my phones, outgoing calls will randomly fail. The problem goes away when I disable SRTP. I only work over TLS.
Incoming calls work reliably with or without SRTP.

How do you suggest debugging this?
I tried setting up a fresh instance of FS but the issue persists.
Now. It should be noted that calls fail sensibly more often when I have more than one account registered on the same server with the same device.

Any suggestion is welcome. Have you experienced this? 

I?m running FreeSWITCH Version 1.6.5 on Debian. My phone in this case is a Yealink SIP-T46G running firmware 28.80.0.95.

E
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From sharigosrl at gmail.com  Fri Feb 26 01:24:41 2016
From: sharigosrl at gmail.com (sharigo roma)
Date: Thu, 25 Feb 2016 23:24:41 +0100
Subject: [Freeswitch-users] discriminate between session answered by users
 and voice mail automatic messages
Message-ID: <CAE6rfk1KG7S8eSgYM_Ydnpx9ix7d4bRQbThQ3OSSaCq_pDa3=w@mail.gmail.com>

Dear freeswitchers,

is there a way to understand if a new session originated from a python
handler has been actually answered by the remote user or by the operator
voice mail with a message like "the customer you are trying to reach is not
available at the moment" ?

The python code looks something like the following:


def handler(session, args):
 ## I do various things.... and than

 session_string = str("sofia/gateway/MY_GW/0039123456789?)
 new_session = freeswitch.Session(session_string)
 if new_session.ready():
    # I get here also in case of automatic voice mail message
 else:
    # I supposed i?d get here in case of automatic ?unreachable? messages?


Thanks in advance
Lorenzo
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From brian at freeswitch.org  Fri Feb 26 02:35:01 2016
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Feb 2016 17:35:01 -0600
Subject: [Freeswitch-users] SRTP breaks my TLS session
In-Reply-To: <84F64031-B943-4671-A0BC-66FC02DF7C51@kavun.ch>
References: <84F64031-B943-4671-A0BC-66FC02DF7C51@kavun.ch>
Message-ID: <CAEJMVkCsxNUfWNBAkJQp5k86Nb+1=vFBj13XKEc+ad1_Hst8og@mail.gmail.com>

Sounds like there may be a bug in the yealink if packet size is of issue on
TCP/TLS connections.

On Thu, Feb 25, 2016 at 5:13 PM, Emrah <lists at kavun.ch> wrote:

> Hello list,
> I thought I had solved this issue by reducing my codec list  to a minimum,
> but it still persists, unfortunately. This was to reduce the TLS packet
> size.
> Anytime I enable SRTP on my phones, outgoing calls will randomly fail. The
> problem goes away when I disable SRTP. I only work over TLS.
> Incoming calls work reliably with or without SRTP.
>
> How do you suggest debugging this?
> I tried setting up a fresh instance of FS but the issue persists.
> Now. It should be noted that calls fail sensibly more often when I have
> more than one account registered on the same server with the same device.
>
> Any suggestion is welcome. Have you experienced this?
>
> I?m running FreeSWITCH Version 1.6.5 on Debian. My phone in this case is
> a Yealink SIP-T46G running firmware 28.80.0.95.
>
> E
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

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 lmorley at neny.cslimits.net  Fri Feb 26 03:09:26 2016
From: lmorley at neny.cslimits.net (Larry Morley)
Date: Thu, 25 Feb 2016 19:09:26 -0500
Subject: [Freeswitch-users] Getting 503 when starting automatically
In-Reply-To: <BN3PR13MB078550D45CAAEDE31D682A60D7A60@BN3PR13MB0785.namprd13.prod.outlook.com>
References: <BN3PR13MB078550D45CAAEDE31D682A60D7A60@BN3PR13MB0785.namprd13.prod.outlook.com>
Message-ID: <CAAkW=Ffrx-QY8N3O1Mz4uOPvetGA4BAWtXF-No41O5SGAK+KGw@mail.gmail.com>

David,

What happens if you either don't start FS via systemd or stop it once the
system's done booting, then start it like, say,

su
PATH=/usr/local/freeswitch/bin:$PATH
freeswitch -nc

Note that you'll likely want to keep a record of what the permissions on
your FS files and directories are BEFORE doing this.

Does it work then? Can you see from the console where the problem is?

I'm nearly positive that I've chased this same error myself before.
Unfortunately, I can't recall what the root cause or the fix was.

Good luck!

- Larry
On Feb 25, 2016 16:30, <freeswitch.opencode at spamgourmet.com> wrote:

> On Thu, 2016-02-25 at 21:26:09 +0300, Bote wrote:
> > A 503 error, huh? Weird. Check for permissions problems.
>
> > My guess is that you are building as "root" so when you first ran it
> > manually all the databases and log files were created with root as owner.
> > But when it boots systemd is starting it as root and then dropping
> > privileges to the freeswitch user as it should. After dropping
> privileges it
> > can't access its config files or open ports necessary to function.
>
> Thanks for the suggestion. I compiled as a normal user using Arch Linux's
> makepkg tool and installed using its pacman tool, so I don't think that's
> the problem. I'm also letting system start freeswitch.service with its
> default user setting (as root).
>
> The weirdest thing is that it does start and connect fine using systemd
> when make a request after boot to have systemd start it (systemctl start
> freeswitch or systemctl restart freeswitch). In both cases, systemd is
> controlling process identity, so I don't think it's a permissions issue. My
> best guess is there's some kind of race condition at boot and if FreeSWITCH
> starts too early, it gets a cached but invalid handle of some sort that
> doesn't allow communicating in or out. This is just a guess though.
>
> 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
>
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From freeswitch at digitaldescent.net  Fri Feb 26 04:18:15 2016
From: freeswitch at digitaldescent.net (Brian Chow)
Date: Thu, 25 Feb 2016 17:18:15 -0800
Subject: [Freeswitch-users] Inbound calls mis-routing routing to
 internal extension with external IP?
In-Reply-To: <bf0ede9ef70640818f0607ceb46f5434@nysolutions.com>
References: <CAA_Md=uAGmnqLkY3g128QcjoBis4VnoTwSUEZpqAws35AwRsHQ@mail.gmail.com>
	<23b1626b3814448b90842a279dc30be5@nysolutions.com>
	<CAA_Md=t4NR-ocAsnvOmD39XGNAexYVZWsWy4taYbiXU7-N1qEg@mail.gmail.com>
	<CAEJMVkC+=U50t2nEYrn9CU1OQnXOUcVXKmpzaB8KF=Gbh7T-BQ@mail.gmail.com>
	<CAA_Md=v=PGrR_0s90SK_ijZO5eScqJYN+Utqu22nwqbZEQXvLA@mail.gmail.com>
	<CAA_Md=sTVGHHmAvtV9EV2OvBuLkdnQ3b9wRaSCMYjBwf71p-yA@mail.gmail.com>
	<bf0ede9ef70640818f0607ceb46f5434@nysolutions.com>
Message-ID: <CAA_Md=tJGhw7_BGz4qFhXAz_dnpy8d7+2L0cBRm6ZaF9jPeHaQ@mail.gmail.com>

Moishe,
  Thanks for double checking this, I hadn't thought to try an extension <->
extension call as it's a private server just for me, however, after
creating another extension, I cannot make internal calls.  It seems, I can
only make outbound calls.

Internal -> External is working.
External -> Internal only works to the FS server, not to the extension.
Internal -> Internal does not work.



On Thu, Feb 25, 2016 at 8:12 AM, Moishe Grunstein <max at nysolutions.com>
wrote:

> Does internal call also go direct to voicemail?
>
> Are you sure you don?t have a *99 before the extension number?
>
>
>
> 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 *Brian Chow
> *Sent:* Thursday, February 25, 2016 11:00 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Inbound calls mis-routing routing to
> internal extension with external IP?
>
>
>
> Ok, in that case, what do I do next to figure out why inbound calls go
> directly to voicemail instead of ringing the extension?
>
> On Feb 25, 2016 7:57 AM, "Brian Chow" <brianchow at digitaldescent.net>
> wrote:
>
> Ok, in that case, what do I look at next to see why it goes straight to
> voicemail instead of ringing my extension?
>
> On Feb 25, 2016 5:52 AM, "Brian West" <brian at freeswitch.org> wrote:
>
> Its just the channel name so don't get tripped up by that, its just named
> with some default data about the session.
>
>
>
> On Wed, Feb 24, 2016 at 11:33 PM, Brian Chow <
> freeswitch at digitaldescent.net> wrote:
>
> Debug console output: https://pastebin.freeswitch.org/24572
>
>
>
> My fqdn only resolves inside the network, but the external-ip is
> configured in FS to use stun, which reports my external ip.
>
>
>
> >Is your endpoint registered? How often is it set to reregister?
>
> Yes, and the zoiper default registration expiry is 3600
>
>
>
> >Does your router have a sip alg? Are the ports opened in your firewall?
>
> > ish, no GUI for it, but I did make sure to disable it from the command
> line (Ubiquiti Edgerouter).  No firewall ports are explicitly open, but
> allow bi-directional with existing states.  I do get 2 way audio when
> calling out, and I also can hear the system forwarding my call in to
> voicemail; the voicemail message and prompts.  I know asterisk is different
> from FS, but my asterisk experience has proven that at least with asterisk,
> port forwarding isn't needed.  Also, unfortunately, I cannot get a static
> ip.
>
>
>
> Thanks!
>
>   -Brian
>
>
>
> On Wed, Feb 24, 2016 at 8:31 PM, Moishe Grunstein <max at nysolutions.com>
> wrote:
>
> It is probably sending call to your extension at domainname, if your
> external ip is your domain name then you will see the external ip.
>
> Is your endpoint registered? How often is it set to reregister?
>
> Does your router have a sip alg? Are the ports opened in your firewall?
>
> Very hard to guess without a log of a call with debug enabled.
>
> sofia global sip trace on
>
>
>
> 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 *Brian Chow
> *Sent:* Wednesday, February 24, 2016 10:00 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Inbound calls mis-routing routing to
> internal extension with external IP?
>
>
>
> Hello all!
>
>   I'm new to freeswitch, so I'm sure this is just a newbie configuration
> error.  Sorry if it's been answered a million times, my google searches
> always just bring up the standard NAT configuration pages.  I've already
> followed the confluence page on configuring NAT.
>
>
>
> My Setup:
>
> Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3
> instance.
>
> SIP Provider: Flowroute
>
>
>
> NAT: extension and freeswitch are on the same network, both of which are
> behind NAT and connecting to flowroute.  I configured external sip and rtp
> to use stun entries.  External profile is using $${external_rtp/sip_ip} for
> ext-rtp/sip respectively.
>
>
>
> I have one DID configured to go directly to my one extension.  My
> extension can register just fine.  My extension can dial out.  When I call
> my cell, the call connects and I have 2 way audio.  When I dial my DID from
> my cell, I can see the call hitting the FS server, but instead of ringing
> my extension, it goes straight to my extensions voicemail (which I can just
> fine).
>
>
>
> When I look the at the console, it appears (sorry if this is wrong, I'm
> only a day into free switch) that FS is attempting to route the call to my
> extension...@ my external ip?
>
>
>
> I see this line in the console:  2016-02-24 18:56:10.453983 [NOTICE]
> switch_channel.c:1101 New Channel sofia/internal/<extension>@
> *<externalip>*:46072
>
>
>
>
>
> Shouldn't that read sofia/internal/<extentsion>@<ip_of_extension>  ?
>
>
>
> Sofia Status says:
>
>
>
>                      Name        Type
>  Data  State
>
>
> =================================================================================================
>
>             external-ipv6   profile
> sip:mod_sofia@[::1]:5080            RUNNING (0)
>
>                  external      profile
> sip:mod_sofia@<external_ip>:5080     RUNNING (0)
>
> external::<uuid>          gateway                      sip:<flowrouteid>@
> sip.flowroute.com          REGED
>
>             internal-ipv6    profile
> sip:mod_sofia@[::1]:5060            RUNNING (0)
>
>                  internal       profile
> sip:mod_sofia@<internal_ip>:5060    RUNNING (0)
>
>
> =================================================================================================
>
>
>
> If I'm completely off base here, can anyone recommend where I can start
> looking to change/troubleshoot the issue?  I feel like it's just me missing
> something, I just can't determine what that might be.
>
>
>
> Thanks,
>
>   -Brian
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
>
> 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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Fri Feb 26 04:50:28 2016
From: max at nysolutions.com (Moishe Grunstein)
Date: Fri, 26 Feb 2016 01:50:28 +0000
Subject: [Freeswitch-users] Inbound calls mis-routing routing to
 internal extension with external IP?
In-Reply-To: <CAA_Md=tJGhw7_BGz4qFhXAz_dnpy8d7+2L0cBRm6ZaF9jPeHaQ@mail.gmail.com>
References: <CAA_Md=uAGmnqLkY3g128QcjoBis4VnoTwSUEZpqAws35AwRsHQ@mail.gmail.com>
	<23b1626b3814448b90842a279dc30be5@nysolutions.com>
	<CAA_Md=t4NR-ocAsnvOmD39XGNAexYVZWsWy4taYbiXU7-N1qEg@mail.gmail.com>
	<CAEJMVkC+=U50t2nEYrn9CU1OQnXOUcVXKmpzaB8KF=Gbh7T-BQ@mail.gmail.com>
	<CAA_Md=v=PGrR_0s90SK_ijZO5eScqJYN+Utqu22nwqbZEQXvLA@mail.gmail.com>
	<CAA_Md=sTVGHHmAvtV9EV2OvBuLkdnQ3b9wRaSCMYjBwf71p-yA@mail.gmail.com>
	<bf0ede9ef70640818f0607ceb46f5434@nysolutions.com>
	<CAA_Md=tJGhw7_BGz4qFhXAz_dnpy8d7+2L0cBRm6ZaF9jPeHaQ@mail.gmail.com>
Message-ID: <62e0a3e6c4e14d40b15aedb1d3e1b3b6@nysolutions.com>

Can you show a log of a internal call?
Are the phones showing registered? Show registrations
Did you by any chance add your ip range to the ACL.


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 Brian Chow
Sent: Thursday, February 25, 2016 8:18 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP?

Moishe,
  Thanks for double checking this, I hadn't thought to try an extension <-> extension call as it's a private server just for me, however, after creating another extension, I cannot make internal calls.  It seems, I can only make outbound calls.

Internal -> External is working.
External -> Internal only works to the FS server, not to the extension.
Internal -> Internal does not work.



On Thu, Feb 25, 2016 at 8:12 AM, Moishe Grunstein <max at nysolutions.com<mailto:max at nysolutions.com>> wrote:
Does internal call also go direct to voicemail?
Are you sure you don?t have a *99 before the extension number?

Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650<tel:212.400.7650> 888.IPPBX.US<http://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> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Brian Chow
Sent: Thursday, February 25, 2016 11:00 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP?


Ok, in that case, what do I do next to figure out why inbound calls go directly to voicemail instead of ringing the extension?
On Feb 25, 2016 7:57 AM, "Brian Chow" <brianchow at digitaldescent.net<mailto:brianchow at digitaldescent.net>> wrote:

Ok, in that case, what do I look at next to see why it goes straight to voicemail instead of ringing my extension?
On Feb 25, 2016 5:52 AM, "Brian West" <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:
Its just the channel name so don't get tripped up by that, its just named with some default data about the session.

On Wed, Feb 24, 2016 at 11:33 PM, Brian Chow <freeswitch at digitaldescent.net<mailto:freeswitch at digitaldescent.net>> wrote:
Debug console output: https://pastebin.freeswitch.org/24572

My fqdn only resolves inside the network, but the external-ip is configured in FS to use stun, which reports my external ip.

>Is your endpoint registered? How often is it set to reregister?
Yes, and the zoiper default registration expiry is 3600

>Does your router have a sip alg? Are the ports opened in your firewall?
> ish, no GUI for it, but I did make sure to disable it from the command line (Ubiquiti Edgerouter).  No firewall ports are explicitly open, but allow bi-directional with existing states.  I do get 2 way audio when calling out, and I also can hear the system forwarding my call in to voicemail; the voicemail message and prompts.  I know asterisk is different from FS, but my asterisk experience has proven that at least with asterisk, port forwarding isn't needed.  Also, unfortunately, I cannot get a static ip.

Thanks!
  -Brian

On Wed, Feb 24, 2016 at 8:31 PM, Moishe Grunstein <max at nysolutions.com<mailto:max at nysolutions.com>> wrote:
It is probably sending call to your extension at domainname, if your external ip is your domain name then you will see the external ip.
Is your endpoint registered? How often is it set to reregister?
Does your router have a sip alg? Are the ports opened in your firewall?
Very hard to guess without a log of a call with debug enabled.
sofia global sip trace on

Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650<tel:212.400.7650> 888.IPPBX.US<http://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> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Brian Chow
Sent: Wednesday, February 24, 2016 10:00 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP?

Hello all!
  I'm new to freeswitch, so I'm sure this is just a newbie configuration error.  Sorry if it's been answered a million times, my google searches always just bring up the standard NAT configuration pages.  I've already followed the confluence page on configuring NAT.

My Setup:
Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3 instance.
SIP Provider: Flowroute

NAT: extension and freeswitch are on the same network, both of which are behind NAT and connecting to flowroute.  I configured external sip and rtp to use stun entries.  External profile is using $${external_rtp/sip_ip} for ext-rtp/sip respectively.

I have one DID configured to go directly to my one extension.  My extension can register just fine.  My extension can dial out.  When I call my cell, the call connects and I have 2 way audio.  When I dial my DID from my cell, I can see the call hitting the FS server, but instead of ringing my extension, it goes straight to my extensions voicemail (which I can just fine).

When I look the at the console, it appears (sorry if this is wrong, I'm only a day into free switch) that FS is attempting to route the call to my extension...@ my external ip?

I see this line in the console:  2016-02-24 18:56:10.453983 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/<extension>@<externalip>:46072


Shouldn't that read sofia/internal/<extentsion>@<ip_of_extension>  ?

Sofia Status says:

                     Name        Type                                      Data  State
=================================================================================================
            external-ipv6   profile                   sip:mod_sofia@[::1]:5080            RUNNING (0)
                 external      profile           sip:mod_sofia@<external_ip>:5080<sip:mod_sofia@%3cexternal_ip%3e:5080>     RUNNING (0)
external::<uuid>          gateway                      sip:<flowrouteid>@sip.flowroute.com<http://sip.flowroute.com>          REGED
            internal-ipv6    profile                   sip:mod_sofia@[::1]:5060            RUNNING (0)
                 internal       profile            sip:mod_sofia@<internal_ip>:5060<sip:mod_sofia@%3cinternal_ip%3e:5060>    RUNNING (0)
=================================================================================================

If I'm completely off base here, can anyone recommend where I can start looking to change/troubleshoot the issue?  I feel like it's just me missing something, I just can't determine what that might be.

Thanks,
  -Brian

_________________________________________________________________________
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>
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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



--

Brian West
brian at freeswitch.org<mailto:brian at freeswitch.org>

[http://billing.freeswitch.org/templates/default/img/whmcslogo.png]

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

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

T:+19184209001<tel:%2B19184209001> | F:+19184209002<tel:%2B19184209002> | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest

_________________________________________________________________________
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
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From italo at freeswitch.org  Fri Feb 26 05:00:49 2016
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Thu, 25 Feb 2016 23:00:49 -0300
Subject: [Freeswitch-users] discriminate between session answered by
 users and voice mail automatic messages
In-Reply-To: <CAE6rfk1KG7S8eSgYM_Ydnpx9ix7d4bRQbThQ3OSSaCq_pDa3=w@mail.gmail.com>
References: <CAE6rfk1KG7S8eSgYM_Ydnpx9ix7d4bRQbThQ3OSSaCq_pDa3=w@mail.gmail.com>
Message-ID: <CABMX_w9f0uEAda4W5sEt=4W_NZBgKURBFN6K6XisYxteakhu7Q@mail.gmail.com>

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

On Thu, Feb 25, 2016 at 7:24 PM, sharigo roma <sharigosrl at gmail.com> wrote:

> Dear freeswitchers,
>
> is there a way to understand if a new session originated from a python
> handler has been actually answered by the remote user or by the operator
> voice mail with a message like "the customer you are trying to reach is not
> available at the moment" ?
>
> The python code looks something like the following:
>
>
> def handler(session, args):
>  ## I do various things.... and than
>
>  session_string = str("sofia/gateway/MY_GW/0039123456789?)
>  new_session = freeswitch.Session(session_string)
>  if new_session.ready():
>     # I get here also in case of automatic voice mail message
>  else:
>     # I supposed i?d get here in case of automatic ?unreachable? messages?
>
>
> Thanks in advance
> Lorenzo
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From italo at freeswitch.org  Fri Feb 26 05:05:30 2016
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Thu, 25 Feb 2016 23:05:30 -0300
Subject: [Freeswitch-users] mod_distributor / gwlist down
In-Reply-To: <CAEzwe14uLCk+MvyQ8mNuixR1Gy13D8ox+TKbOTt9ucQEWdePsw@mail.gmail.com>
References: <CAEzwe14uLCk+MvyQ8mNuixR1Gy13D8ox+TKbOTt9ucQEWdePsw@mail.gmail.com>
Message-ID: <CABMX_w-_pyVGKMDi9LV8phGOo_Gd92qckn+Qqsr3E=Y6=m3i2g@mail.gmail.com>

you missed the profile keywork in your dialplan:

sofia PROFILE internal internal_trk_1 gwlist down

On Wed, Feb 24, 2016 at 1:38 PM, Dominique Jeannerod <
dominique.jeannerod at interact-iv.com> wrote:

> Hello,
>
> i'm currently working with mod_distributor, to load balance calls to
> gateways.
> The load balancing works well, and it's rather simple to configure, no
> problem.
>
> I'm trying to automatically exclude down gateways from the load balancing,
> which seems quite simple, too, when reading the mod_distributor doc... but
> i can't make it work.
>
> I'm using freeswitch 1.6.6.
> I configured a profile internal_trk_1, and 2 gateways (svc-isr-fr-301, and
> non_existent_gw), with ping option.
> One of the gateways, called (non_existent_gw) is down, and detected down
> by sofia
>
> When I check the status of the gateways, the result is ok :
> sofia profile internal_trk_1 gwlist down
> non_existent_gw
>
> When I check with distributor command in the cli, it's also ok :
> expand distributor I_MUT_SIPGTW ${sofia profile internal_trk_1 gwlist down}
> svc-isr-fr-301
>
> But I can't make it work inside the dialplan. I tested using the 2
> syntaxes listed in
> https://freeswitch.org/confluence/display/FREESWITCH/mod_distributor :
>
> <action application="bridge" data="sofia/gateway/${distributor
> I_MUT_SIPGTW ${sofia internal_trk_1 gwlist down}}/${destination_number}"/>
>
> <action application="bridge" data="sofia/gateway\${expand(distributor
> I_MUT_SIPGTW \${sofia(internal_trk_1 gwlist
> down)})}/${destination_number}"/>
>
> The dead gateway still gets used by the load balancer.
>
> What I am doing wrong ?
>
> Best regards
>
>
> Dominique Jeannerod
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From andrew.keil at visytel.com  Fri Feb 26 05:33:08 2016
From: andrew.keil at visytel.com (Andrew Keil)
Date: Fri, 26 Feb 2016 02:33:08 +0000
Subject: [Freeswitch-users] Re- End Lua script after HangupHook handled
 without all the extra code to handle the return to the function
Message-ID: <SY3PR01MB015646922642A69251933890F3A70@SY3PR01MB0156.ausprd01.prod.outlook.com>

To FreeSWITCH Users,

See below for a sample template for a Lua Service Script running inside FreeSWITCH.

The issue I have is fairly straightforward.

I need a function to run when hangup is detected (ie. at the end of the call) however I understand this must not delay ending the script.  This function is CleanUp().  Then I would like the service to end.

The problem I am having is if the caller hangs up during the playback of "intro.wav" (as shown inside the MainService() function below), then the code jumps to the myHangupHook which calls CleanUp() perfectly, the issue is once CleanUp() is complete I would like the Lua script to end there and then (ie. at the bottom of CleanUp()).  What actually happens is it returns to MainService() and continues to try and play "info.wav", unless I either check for session:ready() everywhere or add a goto as shown below under each streamFile() function call.

My aim is to reduce extra code and to make the Lua script simpler and easier to read.  Also I would like to try and avoid goto statements, which I know can be done with if (session:ready()) etc....

So is there a way to stop a Lua script running inside FreeSWITCH cleanly?  I have tried the os.exit() this is barred from use by FreeSWITCH.  I have also tried session:destroy() which crashes FreeSWITCH (version 1.6.5 on CentOS 6.7, CentOS 7 and windows) 100% of the time!

I could look further into the Lua additions done by the FreeSWITCH team in the source code, however if someone has already solved this then that would be the best solution.

FYI: Obviously the script below is simple, however I am sure that you understand if the script was complicated having to use "if (session:ready()) then ...." or "if (not session:ready()) then goto HANGUPEXIT end" makes the code ugly.

Thanks in advance,

Andrew Keil
Visytel Pty Ltd



------------------------------------------------------------------------------  Sample Lua Service -----------------------------------------------------------------------

-- Lua template for FreeSWITCH service
-- By: Andrew Keil (Visytel Pty Ltd)
-- Email: support at visytel.com

-- Setup script wide variables here

function PreAnswer()
                freeswitch.consoleLog("INFO", "PRE ANSWER SECTION\n");
                -- Add your pre answer code from here

                -- End of your pre answer code
                freeswitch.consoleLog("INFO", "PRE ANSWER SECTION COMPLETE\n");
end

function AnswerCaller()
                session:answer()
                session:sleep(1000)
end

function MainService()
                freeswitch.consoleLog("INFO", "MAIN SERVICE SECTION\n");
                if (session:ready()) then
                                -- Note (1): If you wish to end the call then simply use: goto ENDSERVICE
                                -- Note (2): To terminate the service sooner when HANGUP is detected use: if (not session:ready()) then goto HANGUPEXIT end
                                -- Add your main service code from here               (caller would have been answered)

                                session:streamFile("intro.wav")
                                if (not session:ready()) then goto HANGUPEXIT end
                                session:streamFile("info.wav")
                                if (not session:ready()) then goto HANGUPEXIT end
                                session:streamFile("outro.wav")
                                if (not session:ready()) then goto HANGUPEXIT end

                                -- End of your main service code
                end
                ::ENDSERVICE::
                if (session:ready()) then
                                -- End of service so hangup
                                session:hangup()  -- Should automatically jump to CleanUp() via hangup handler if caller still online at this stage
                end
                goto END
                ::HANGUPEXIT::
                freeswitch.consoleLog("INFO", "END OF SERVICE (HANGUP DETECTED)\n");
                ::END::
                freeswitch.consoleLog("INFO", "MAIN SERVICE SECTION COMPLETE\n");
end

function CleanUp()
                freeswitch.consoleLog("INFO", "CLEANUP SECTION\n");
                -- Add your cleanup code from here (caller would have been disconnected)

                -- End of your cleanup code
                freeswitch.consoleLog("INFO", "CLEANUP SECTION COMPLETE\n");
end

function myHangupHook(s, status, arg)
                session:hangup()
                CleanUp() -- Run CleanUp function now since the caller has disconnected
end

-- Setup Hangup event handler here
v_hangup = "HANGUP"
session:setHangupHook("myHangupHook", "v_hangup")

-- Call service functions in order
PreAnswer()
AnswerCaller()
MainService()
-- End of Lua service


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From mario_fs at mgtech.com  Fri Feb 26 08:37:41 2016
From: mario_fs at mgtech.com (Mario G)
Date: Thu, 25 Feb 2016 21:37:41 -0800
Subject: [Freeswitch-users] SRTP breaks my TLS session
In-Reply-To: <CAEJMVkCsxNUfWNBAkJQp5k86Nb+1=vFBj13XKEc+ad1_Hst8og@mail.gmail.com>
References: <84F64031-B943-4671-A0BC-66FC02DF7C51@kavun.ch>
	<CAEJMVkCsxNUfWNBAkJQp5k86Nb+1=vFBj13XKEc+ad1_Hst8og@mail.gmail.com>
Message-ID: <88ACC7C7-589C-4CC8-A146-3DBF31B84A2E@mgtech.com>

You may want to run a pcap trace on the Yealink. It?s under settings->Configuration. Start/test/export.

> On Feb 25, 2016, at 3:35 PM, Brian West <brian at freeswitch.org> wrote:
> 
> Sounds like there may be a bug in the yealink if packet size is of issue on TCP/TLS connections.
> 
> On Thu, Feb 25, 2016 at 5:13 PM, Emrah <lists at kavun.ch <mailto:lists at kavun.ch>> wrote:
> Hello list,
> I thought I had solved this issue by reducing my codec list  to a minimum, but it still persists, unfortunately. This was to reduce the TLS packet size.
> Anytime I enable SRTP on my phones, outgoing calls will randomly fail. The problem goes away when I disable SRTP. I only work over TLS.
> Incoming calls work reliably with or without SRTP.
> 
> How do you suggest debugging this?
> I tried setting up a fresh instance of FS but the issue persists.
> Now. It should be noted that calls fail sensibly more often when I have more than one account registered on the same server with the same device.
> 
> Any suggestion is welcome. Have you experienced this? 
> 
> I?m running FreeSWITCH Version 1.6.5 on Debian. My phone in this case is a Yealink SIP-T46G running firmware 28.80.0.95.
> 
> E
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> Brian West
> brian at freeswitch.org <mailto:brian at freeswitch.org>
> 
> Twitter: @FreeSWITCH , @briankwest
> http://www.freeswitchbook.com <http://www.freeswitchbook.com/>
> http://www.freeswitchcookbook.com <http://www.freeswitchcookbook.com/>
> 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 lists at kavun.ch  Fri Feb 26 12:36:49 2016
From: lists at kavun.ch (Emrah)
Date: Fri, 26 Feb 2016 10:36:49 +0100
Subject: [Freeswitch-users] SRTP breaks my TLS session
In-Reply-To: <88ACC7C7-589C-4CC8-A146-3DBF31B84A2E@mgtech.com>
References: <84F64031-B943-4671-A0BC-66FC02DF7C51@kavun.ch>
	<CAEJMVkCsxNUfWNBAkJQp5k86Nb+1=vFBj13XKEc+ad1_Hst8og@mail.gmail.com>
	<88ACC7C7-589C-4CC8-A146-3DBF31B84A2E@mgtech.com>
Message-ID: <93C84919-0D70-4DD3-BB86-46DE87492F9C@kavun.ch>

Thanks for this. 

This isn?t just a yealink thing. I?ve encountered sporadic issues with soft phones and other desk phones as well.

I didn?t use the PCap capture feature because I had assumed it would give me a bunch of TLS packets. I?ll test that and revert back.

How can we explain that I have more calls failing if I register multiple accounts?

Emrah

> On Feb 26, 2016, at 6:37 AM, Mario G <mario_fs at mgtech.com> wrote:
> 
> You may want to run a pcap trace on the Yealink. It?s under settings->Configuration. Start/test/export.
> 
>> On Feb 25, 2016, at 3:35 PM, Brian West <brian at freeswitch.org <mailto:brian at freeswitch.org>> wrote:
>> 
>> Sounds like there may be a bug in the yealink if packet size is of issue on TCP/TLS connections.
>> 
>> On Thu, Feb 25, 2016 at 5:13 PM, Emrah <lists at kavun.ch <mailto:lists at kavun.ch>> wrote:
>> Hello list,
>> I thought I had solved this issue by reducing my codec list  to a minimum, but it still persists, unfortunately. This was to reduce the TLS packet size.
>> Anytime I enable SRTP on my phones, outgoing calls will randomly fail. The problem goes away when I disable SRTP. I only work over TLS.
>> Incoming calls work reliably with or without SRTP.
>> 
>> How do you suggest debugging this?
>> I tried setting up a fresh instance of FS but the issue persists.
>> Now. It should be noted that calls fail sensibly more often when I have more than one account registered on the same server with the same device.
>> 
>> Any suggestion is welcome. Have you experienced this? 
>> 
>> I?m running FreeSWITCH Version 1.6.5 on Debian. My phone in this case is a Yealink SIP-T46G running firmware 28.80.0.95.
>> 
>> E
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> 
>> 
>> 
>> -- 
>> Brian West
>> brian at freeswitch.org <mailto:brian at freeswitch.org>
>> 
>> Twitter: @FreeSWITCH , @briankwest
>> http://www.freeswitchbook.com <http://www.freeswitchbook.com/>
>> http://www.freeswitchcookbook.com <http://www.freeswitchcookbook.com/>
>> 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 <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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 tarantul at gmail.com  Fri Feb 26 13:04:34 2016
From: tarantul at gmail.com (Nick 'tarantul' Novikov)
Date: Fri, 26 Feb 2016 13:04:34 +0300
Subject: [Freeswitch-users] no audio after hold
In-Reply-To: <CAK+V7Rwfy3+fRsr49dmbfWeZ6fDhiaDdJMWXx5T6KD1XoJo7fA@mail.gmail.com>
References: <CAK+V7RxSdxaSLAE3iwUqSpQkTorED0X24PpG0YL3tEbOgah3bw@mail.gmail.com>
	<CALXCt0rFM5xkK5JiwcqE5pj2pgFVkTaKZ8-qxcuqY4N1GGJLrA@mail.gmail.com>
	<CAEJMVkCRrBU8DC=PFua2yb_N48-xBR3r1v9YSaCG5Ss7sV4=_w@mail.gmail.com>
	<CAK+V7RygPe5Bi-xzabjSdgovbvwo++k=X+FwZ0nqNQaeGkKYFQ@mail.gmail.com>
	<CAEJMVkDMFN3hPgSujiSvhk6ZpS-H+jmod468H9YKrW+XH7M+pQ@mail.gmail.com>
	<CAK+V7Rwfy3+fRsr49dmbfWeZ6fDhiaDdJMWXx5T6KD1XoJo7fA@mail.gmail.com>
Message-ID: <CAK+V7RxjN30MKCpJX6o91T-vD1KpS874NcC3OeoXDbcWqronNA@mail.gmail.com>

Do you have any news about this bug?
Can I help?

On Wed, Feb 17, 2016 at 11:32 PM, Nick 'tarantul' Novikov <
tarantul at gmail.com> wrote:

> Hello!
>
> Done.
> https://freeswitch.org/jira/browse/FS-8840
>
> On Wed, Feb 17, 2016 at 5:31 PM, Brian West <brian at freeswitch.org> wrote:
>
>> bug reports go on JIRA please.
>>
>> On Wed, Feb 17, 2016 at 4:03 AM, Nick 'tarantul' Novikov <
>> tarantul at gmail.com> wrote:
>>
>>> Hello!
>>>
>>> Log file is large, I can't paste it in FS pastebin.
>>> You may download gzip log file here
>>> https://yadi.sk/d/pLbEHi_LoyWsS
>>>
>>> On Tue, Feb 16, 2016 at 7:11 PM, Brian West <brian at freeswitch.org>
>>> wrote:
>>>
>>>> Your logs are incomplete, need full debug logs.  Please don't filter
>>>> them excessively and try on master please.
>>>>
>>>> On Tue, Feb 16, 2016 at 10:06 AM, Giovanni Maruzzelli <
>>>> gmaruzz at gmail.com> wrote:
>>>>
>>>>> Please pastebin a complete sip trace (from fs-cli: sofia global
>>>>> siptrace on) and then put here the link to pastebin.
>>>>> Il 16/Feb/2016 16:29, "Nick 'tarantul' Novikov" <tarantul at gmail.com>
>>>>> ha scritto:
>>>>>
>>>>>> Hello
>>>>>>
>>>>>> I have some problem with sound after hold usage. Sometime the sound
>>>>>> disappear after callcenter agent shift unhold.For callcenter agents we use
>>>>>> sip.js (version 0.7.2).
>>>>>> In freeswitch logs I see strange lines (
>>>>>> https://pastebin.freeswitch.org/24557).
>>>>>> Can anyone explain, what happened with audio and how to fix it?
>>>>>>
>>>>>>
>>>>>> --
>>>>>> tarantul
>>>>>> Dios es Amor
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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
>>>>
>>>> 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
>>>>
>>>
>>>
>>>
>>> --
>>> tarantul
>>> Dios es Amor
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>> 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
>>
>
>
>
> --
> tarantul
> Dios es Amor
>



-- 
tarantul
Dios es Amor
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From lists at kavun.ch  Fri Feb 26 14:24:25 2016
From: lists at kavun.ch (Emrah)
Date: Fri, 26 Feb 2016 12:24:25 +0100
Subject: [Freeswitch-users] Joining several FS servers in different
	regions
In-Reply-To: <20160224152350.GA30297@blomma.liberationtech.net>
References: <20160224152350.GA30297@blomma.liberationtech.net>
Message-ID: <50E27BAC-AA7E-4E20-88CC-648CD0DEC6D0@kavun.ch>

Hi there,
I was recently confronted to a similar challenge. 
In my case, there was an additional use case where Bob could have multiple phones registered simultaneously both on the North American server and the EU server.
I strive to involve as little components as possible to avoid unnecessary breaking points. So what I came up with is a very basic but working solution.
I modified my dial string so that user/1000 at domain would also ring the alternative server in bypass media mode.
I also share the same directory files and dial plan files that I host on a centralized storage space.
My gateways are set up strategically to route the EU server to the EU POPs and the U.S. server to the NA POPs. That is for my providers who have multiple POPs distributed geographically.

With the set up as described here, both servers are active and passive at the same time in the cluster. And it can easily be expanded. To my knowledge, it?s as optimized as it can get.

Best,
Emrah
> On Feb 24, 2016, at 4:23 PM, Oivvio Polite <mylists at polite.se> wrote:
> 
> 
> I'm working on a Saas service that will connect WebRTC (with SIP or Verto for
> signaling) clients to one another and to PSTN. I'm aware of other such
> services, but this is tailored to a specific niche. I also want WebRTC
> media to flow through FS, rather than P2P, so that I can record it.
> 
> The primary markets are Europe and North America. To keep latency low I need
> to have servers on both continents. Later on I might want to add more
> regions. I'm not too worried about high availability or high loads. 
> 
> The clients can keep track of which FS is closest to them and register
> with that. But this still leaves me with a couple of questions. 
> 
> 
> 
> 1. Maintaining a single user database
> =====================================
> 
> What's the simplest way of maintaining one joint user database? (User
> directory in FS parlance). 
> 
> The first thing that comes to mind is to generate the xml dynamically
> from a central database, and force as `reloadxml` on all servers every
> time there's an update. The communication between central database and
> FS servers could be via RabbitMQ.  mod_xml_curl could also be a part of
> this I guess.
> 
> How does that sound? Is there an other simpler way of achiving my goal
> that I'm not seeing?
> 
> 
> 
> 2. Keeping track of where a user i currently registered
> =======================================================
> 
> Let's say I have a client Alice who's registered with a FS server in
> North America and a client Bob who has registered wit a FS server in Europe.
> 
> Now Alice tries to call Bob. How does the North America server know that
> Bob is currently registered with the Europe server and that the call
> should be routed through there?
> 
> For this second problem I don't even have a tentative solution so I'm
> curious to hear any ideas.
> 
> 3. SIP Trunks and geography
> ===========================
> 
> If Alice is in North America and wants to call a PSTN endpoint in North
> America I should obviously be using a SIP trunk in North America, since
> going over the Atlantic twice will add a lot of latency.
> 
> But what about when Alice want to call a PSTN endpoint in Europe? Is it
> better to use a SIP endpoint in Europe or in North America?
> 
> Thanks in advance Oivvio
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Alexey.Gorbachev at leomax.ru  Fri Feb 26 12:47:47 2016
From: Alexey.Gorbachev at leomax.ru (Gorbachev Alexey)
Date: Fri, 26 Feb 2016 09:47:47 +0000
Subject: [Freeswitch-users] FreeSwitch second recovery
Message-ID: <d59189223352415e9daa9dd8c267a811@COD-EXC006.tts.loc>

I installed FreeSwitch cluster according to official manual -https://wiki.freeswitch.org/wiki/Freeswitch_HA

And it works, when I power off first node current calls successfully move to second node and voice disappears only for 3 second.

The problem is when I power on first node, server starts FreeSwitch, and FreeSwitch during start up clears calls in database and of cause I can't move current calls back again to first node.

Can I move current calls between servers without interruption again? Thank you.


Best regards,
Gorbachev Alexey,
System administrator Linux
LEOMAX Group

Tel: +7(495)967-7797, ext. 4119,
Cell: +7(926)982-9520
Mail to: alexey.gorbachev at leomax.ru

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From Adam.Seeliger at qsc.de  Fri Feb 26 15:37:07 2016
From: Adam.Seeliger at qsc.de (Seeliger, Adam)
Date: Fri, 26 Feb 2016 12:37:07 +0000
Subject: [Freeswitch-users] all-reg-options-ping and tls issue
In-Reply-To: <6D3DEA47-DECB-4509-A327-B8DFEC1C1A83@kavun.ch>
References: <2E67ADAE1D3582409C90EBAE8C64C5070149F6B3@QSCDEMXP01a.ONE4ALL.LAN>
	<6D3DEA47-DECB-4509-A327-B8DFEC1C1A83@kavun.ch>
Message-ID: <2E67ADAE1D3582409C90EBAE8C64C507122D35C1@QSCDEMXP01a.ONE4ALL.LAN>

Hi and thanks for the feedback,

sry that I did not respond for a long time.

I already use:
<variable name=?sip-force-contact? value="NDLB-connectile-dysfunction-2.0"/>

I also tested all mentioned params below, nothing works.

When I register a User via TLS FreeSWITCH does not even try to ping the user.
I turned sofia global siptrace on and watched the flow:

User                                        Server
13:09:33.311446:             REGISTER [TLS] ->
13:09:33.312552:             <- 401 UNAUTHORIZED [TLS]
13:09:33.331948:             REGISTER (AUTH) [TLS]  ->
13:09:33.336959:             <- 200 OK [TLS]
Nothing happens
2016-02-26 13:10:00.619525 [WARNING] sofia.c:5769 Sip user 'user at host' is now Unreachable
2016-02-26 13:10:00.619525 [WARNING] sofia.c:5780 Expire sip user 'user at host' due to options failure

When I REGISTER the User via UDP FreeSWITCH starts to ping (OPTIONS) the user as soon as he is registered.

Is there any way to force FreeSWITCH to send OPTIONs in both, udp and tls, depending on the registration?

Thanks in advance,
Adam

Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Emrah
Gesendet: Freitag, 29. Januar 2016 09:25
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] all-reg-options-ping and tls issue

Hi!
This is interesting. I experienced something rather similar where calls would drop because FS would timeout on certain packets sent over UDP instead of TLS.
I assume you mean FS exits with port 5060 instead of port 5061? Because the port on the remote end should be dynamically set.
I found out that in my case, what works best even with TLS, is to use:
<variable name=?sip-force-contact? value="NDLB-connectile-dysfunction-2.0"/>
This goes as far as it can to lay out the path to contacting the client with all consideration in regards to NAT and dynamic ports.
Not sure if it will help you. I?ve personally disabled options-ping an let my clients deal with keep-alive instead.

You could also look into:
    <!-- add a ;received="<ip>:<port>" to the contact when replying to register for nat handling -->
<!? <param name="NDLB-received-in-nat-reg-contact" value="true"/> -->
<!? <param name="bind-params" value="transport=tcp,udp,tls"/>  -->
<!-- additional bind parameters for TLS -->
    <param name=?tls-bind-params" value="transport=tls"/>

I?ll leave it up to you to investigate those options more in details on the FS documentation.

Please keep us posted!

E
On Jan 28, 2016, at 11:48 AM, Seeliger, Adam <Adam.Seeliger at qsc.de<mailto:Adam.Seeliger at qsc.de>> wrote:

Hi all,

I have a problem, when I enable TLS and register a phone using TLS on Port 5061.
FreeSWITCH still tries to ?ping? the phone using Port 5060 using UDP, which is ignored by the phone.
Moments later FreeSWITCH deletes the registration, because ?unregister-on-options-fail? is set to ?true?.

I already figured out, that you can set ?all-reg-options-ping? to ?udp-only?, but this would completely disable this feature for TLS.
Is there any way to ping TLS registered using TLS?

Thanks in advance

- Adam

_________________________________________________________________________
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 s.safarov at gmail.com  Fri Feb 26 15:58:41 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 26 Feb 2016 15:58:41 +0300
Subject: [Freeswitch-users] all-reg-options-ping and tls issue
In-Reply-To: <2E67ADAE1D3582409C90EBAE8C64C507122D35C1@QSCDEMXP01a.ONE4ALL.LAN>
References: <2E67ADAE1D3582409C90EBAE8C64C5070149F6B3@QSCDEMXP01a.ONE4ALL.LAN>
	<6D3DEA47-DECB-4509-A327-B8DFEC1C1A83@kavun.ch>
	<2E67ADAE1D3582409C90EBAE8C64C507122D35C1@QSCDEMXP01a.ONE4ALL.LAN>
Message-ID: <CAHtxdDdy-nq_gZDLKW5bE62-5TQV2u-fK=sSmVD5Y07qv=0Cxw@mail.gmail.com>

If your phone has enabled SIPS uri please disable and use sip+tls.


On Fri, Feb 26, 2016 at 3:37 PM, Seeliger, Adam <Adam.Seeliger at qsc.de>
wrote:

> Hi and thanks for the feedback,
>
>
>
> sry that I did not respond for a long time.
>
>
>
> I already use:
>
> <variable name=?sip-force-contact?
> value="NDLB-connectile-dysfunction-2.0"/>
>
>
>
> I also tested all mentioned params below, nothing works.
>
>
>
> When I register a User via TLS FreeSWITCH does not even try to ping the
> user.
>
> I turned sofia global siptrace on and watched the flow:
>
>
>
> User                                        Server
>
> 13:09:33.311446:             REGISTER [TLS] ->
>
> 13:09:33.312552:             <- 401 UNAUTHORIZED [TLS]
>
> 13:09:33.331948:             REGISTER (AUTH) [TLS]  ->
>
> 13:09:33.336959:             <- 200 OK [TLS]
>
> Nothing happens
>
> 2016-02-26 13:10:00.619525 [WARNING] sofia.c:5769 Sip user 'user at host' is
> now Unreachable
>
> 2016-02-26 13:10:00.619525 [WARNING] sofia.c:5780 Expire sip user
> 'user at host' due to options failure
>
>
>
> When I REGISTER the User via UDP FreeSWITCH starts to ping (OPTIONS) the
> user as soon as he is registered.
>
>
>
> Is there any way to force FreeSWITCH to send OPTIONs in both, udp and tls,
> depending on the registration?
>
>
>
> Thanks in advance,
>
> Adam
>
>
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Emrah
> *Gesendet:* Freitag, 29. Januar 2016 09:25
> *An:* FreeSWITCH Users Help
> *Betreff:* Re: [Freeswitch-users] all-reg-options-ping and tls issue
>
>
>
> Hi!
>
> This is interesting. I experienced something rather similar where calls
> would drop because FS would timeout on certain packets sent over UDP
> instead of TLS.
>
> I assume you mean FS exits with port 5060 instead of port 5061? Because
> the port on the remote end should be dynamically set.
>
> I found out that in my case, what works best even with TLS, is to use:
>
> <variable
> name=?sip-force-contact? value="NDLB-connectile-dysfunction-2.0"/>
>
> This goes as far as it can to lay out the path to contacting the client
> with all consideration in regards to NAT and dynamic ports.
>
> Not sure if it will help you. I?ve personally disabled options-ping an let
> my clients deal with keep-alive instead.
>
>
>
> You could also look into:
>
>     <!-- add a ;received="<ip>:<port>" to the contact when replying to
> register for nat handling -->
>
> <!? <param name="NDLB-received-in-nat-reg-contact" value="true"/> -->
>
> <!? <param name="bind-params" value="transport=tcp,udp,tls"/>  -->
>
> <!-- additional bind parameters for TLS -->
>
>     <param name=?tls-bind-params" value="transport=tls"/>
>
>
>
> I?ll leave it up to you to investigate those options more in details on
> the FS documentation.
>
>
>
> Please keep us posted!
>
>
>
> E
>
> On Jan 28, 2016, at 11:48 AM, Seeliger, Adam <Adam.Seeliger at qsc.de> wrote:
>
>
>
> Hi all,
>
>
>
> I have a problem, when I enable TLS and register a phone using TLS on Port
> 5061.
>
> FreeSWITCH still tries to ?ping? the phone using Port 5060 using UDP,
> which is ignored by the phone.
>
> Moments later FreeSWITCH deletes the registration, because
> ?unregister-on-options-fail? is set to ?true?.
>
>
>
> I already figured out, that you can set ?all-reg-options-ping? to
> ?udp-only?, but this would completely disable this feature for TLS.
>
> Is there any way to ping TLS registered using TLS?
>
>
>
> Thanks in advance
>
>
>
> - Adam
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Adam.Seeliger at qsc.de  Fri Feb 26 16:21:36 2016
From: Adam.Seeliger at qsc.de (Seeliger, Adam)
Date: Fri, 26 Feb 2016 13:21:36 +0000
Subject: [Freeswitch-users] all-reg-options-ping and tls issue
In-Reply-To: <CAHtxdDdy-nq_gZDLKW5bE62-5TQV2u-fK=sSmVD5Y07qv=0Cxw@mail.gmail.com>
References: <2E67ADAE1D3582409C90EBAE8C64C5070149F6B3@QSCDEMXP01a.ONE4ALL.LAN>
	<6D3DEA47-DECB-4509-A327-B8DFEC1C1A83@kavun.ch>
	<2E67ADAE1D3582409C90EBAE8C64C507122D35C1@QSCDEMXP01a.ONE4ALL.LAN>
	<CAHtxdDdy-nq_gZDLKW5bE62-5TQV2u-fK=sSmVD5Y07qv=0Cxw@mail.gmail.com>
Message-ID: <2E67ADAE1D3582409C90EBAE8C64C507122D35E9@QSCDEMXP01a.ONE4ALL.LAN>

Hi,

the phone uses sip+tls.
I test using a snom715, got plenty other phones here, but I guess they will behave the same way.
It really looks like FreeSWITCH is doing something wrong (or is wrongly configured ? if there are any parameters for options ping supporting both, udp and tls)

Regards

Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Sergey Safarov
Gesendet: Freitag, 26. Februar 2016 13:59
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] all-reg-options-ping and tls issue

If your phone has enabled SIPS uri please disable and use sip+tls.


On Fri, Feb 26, 2016 at 3:37 PM, Seeliger, Adam <Adam.Seeliger at qsc.de<mailto:Adam.Seeliger at qsc.de>> wrote:
Hi and thanks for the feedback,

sry that I did not respond for a long time.

I already use:
<variable name=?sip-force-contact? value="NDLB-connectile-dysfunction-2.0"/>

I also tested all mentioned params below, nothing works.

When I register a User via TLS FreeSWITCH does not even try to ping the user.
I turned sofia global siptrace on and watched the flow:

User                                        Server
13:09:33.311446:             REGISTER [TLS] ->
13:09:33.312552:             <- 401 UNAUTHORIZED [TLS]
13:09:33.331948:             REGISTER (AUTH) [TLS]  ->
13:09:33.336959:             <- 200 OK [TLS]
Nothing happens
2016-02-26 13:10:00.619525 [WARNING] sofia.c:5769 Sip user 'user at host' is now Unreachable
2016-02-26 13:10:00.619525 [WARNING] sofia.c:5780 Expire sip user 'user at host' due to options failure

When I REGISTER the User via UDP FreeSWITCH starts to ping (OPTIONS) the user as soon as he is registered.

Is there any way to force FreeSWITCH to send OPTIONs in both, udp and tls, depending on the registration?

Thanks in advance,
Adam

Von: 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>] Im Auftrag von Emrah
Gesendet: Freitag, 29. Januar 2016 09:25
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] all-reg-options-ping and tls issue

Hi!
This is interesting. I experienced something rather similar where calls would drop because FS would timeout on certain packets sent over UDP instead of TLS.
I assume you mean FS exits with port 5060 instead of port 5061? Because the port on the remote end should be dynamically set.
I found out that in my case, what works best even with TLS, is to use:
<variable name=?sip-force-contact? value="NDLB-connectile-dysfunction-2.0"/>
This goes as far as it can to lay out the path to contacting the client with all consideration in regards to NAT and dynamic ports.
Not sure if it will help you. I?ve personally disabled options-ping an let my clients deal with keep-alive instead.

You could also look into:
    <!-- add a ;received="<ip>:<port>" to the contact when replying to register for nat handling -->
<!? <param name="NDLB-received-in-nat-reg-contact" value="true"/> -->
<!? <param name="bind-params" value="transport=tcp,udp,tls"/>  -->
<!-- additional bind parameters for TLS -->
    <param name=?tls-bind-params" value="transport=tls"/>

I?ll leave it up to you to investigate those options more in details on the FS documentation.

Please keep us posted!

E
On Jan 28, 2016, at 11:48 AM, Seeliger, Adam <Adam.Seeliger at qsc.de<mailto:Adam.Seeliger at qsc.de>> wrote:

Hi all,

I have a problem, when I enable TLS and register a phone using TLS on Port 5061.
FreeSWITCH still tries to ?ping? the phone using Port 5060 using UDP, which is ignored by the phone.
Moments later FreeSWITCH deletes the registration, because ?unregister-on-options-fail? is set to ?true?.

I already figured out, that you can set ?all-reg-options-ping? to ?udp-only?, but this would completely disable this feature for TLS.
Is there any way to ping TLS registered using TLS?

Thanks in advance

- Adam

_________________________________________________________________________
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 telishisheer at gmail.com  Fri Feb 26 16:39:03 2016
From: telishisheer at gmail.com (Shisheer Teli)
Date: Fri, 26 Feb 2016 19:09:03 +0530
Subject: [Freeswitch-users] Multiple device registration on single extension
Message-ID: <CACkRGLyUBU2aQyMaXU35betQX95WnJMO0FSwYXVrdr+XKB-kAQ@mail.gmail.com>

Dear Team,

Multiple device registrations on a single extension are not working.

I made changes in internal profile as "<param name="multiple-registrations"
value="true"/>" .

but still not working.

I am using FreeSWITCH (Version 1.6.5 git  64bit) on Linux


-- 
Regards,
Shisheer T
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From mike at jerris.com  Fri Feb 26 16:49:40 2016
From: mike at jerris.com (Michael Jerris)
Date: Fri, 26 Feb 2016 08:49:40 -0500
Subject: [Freeswitch-users] Multiple device registration on single
	extension
In-Reply-To: <CACkRGLyUBU2aQyMaXU35betQX95WnJMO0FSwYXVrdr+XKB-kAQ@mail.gmail.com>
References: <CACkRGLyUBU2aQyMaXU35betQX95WnJMO0FSwYXVrdr+XKB-kAQ@mail.gmail.com>
Message-ID: <CAEwTsAimpqcjcxcCookwWumSCoNDcF7v7vN-u6CyDZqvccvL4w@mail.gmail.com>

Can you give us any more information?

On Friday, February 26, 2016, Shisheer Teli <telishisheer at gmail.com> wrote:

> Dear Team,
>
> Multiple device registrations on a single extension are not working.
>
> I made changes in internal profile as "<param
> name="multiple-registrations" value="true"/>" .
>
> but still not working.
>
> I am using FreeSWITCH (Version 1.6.5 git  64bit) on Linux
>
>
> --
> Regards,
> Shisheer T
>
>
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From Chris.Young at enghouse.com  Fri Feb 26 17:20:00 2016
From: Chris.Young at enghouse.com (Chris Young)
Date: Fri, 26 Feb 2016 14:20:00 +0000
Subject: [Freeswitch-users] Call gets disconnected on write frame error
Message-ID: <a5dbdc1387a84d32b76b6a42d1eb1be9@UK-MAIL-001.edge.local>

Hi all,

Recently, we've been experiencing a problem whereby FreeSWITCH disconnects a call after receiving a frame of audio data with a data length of 0. When this happens we see the following messages in the log:

                [DEBUG] switch_core_io.c:1503 Engaging Write Buffer at 320 bytes to accommodate 0->320
                [ERR] switch_core_io.c:1518 Write Buffer 0 bytes Failed!

and the call gets disconnected. This normally happens when trying to join a conference but we have also seen it when bridging two calls together.

Is the call disconnection by-design or just a side-effect of having received the bad frame of data?

Thanks,
Chris
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From abaci64 at gmail.com  Fri Feb 26 17:29:17 2016
From: abaci64 at gmail.com (Abaci B)
Date: Fri, 26 Feb 2016 09:29:17 -0500
Subject: [Freeswitch-users] Re- End Lua script after HangupHook handled
 without all the extra code to handle the return to the function
In-Reply-To: <SY3PR01MB015646922642A69251933890F3A70@SY3PR01MB0156.ausprd01.prod.outlook.com>
References: <SY3PR01MB015646922642A69251933890F3A70@SY3PR01MB0156.ausprd01.prod.outlook.com>
Message-ID: <CAFmZFiMFsg5JYGE1Ce6=CVdmiUY5+FUVbm5NHFKE_+enPFNC1g@mail.gmail.com>

See
https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:setHangupHook
for a few ways to exit the lua script (error(), return "exit", return
"die", s:destroy("error message")). I personally tried return "exit" but it
seems to me that it only exits the calling function, haven't had a chance
to look further, it's possible that the calling it from the within a
function is different. if you play around and figure out please report back.

On Thu, Feb 25, 2016 at 9:33 PM, Andrew Keil <andrew.keil at visytel.com>
wrote:

> To FreeSWITCH Users,
>
>
>
> See below for a sample template for a Lua Service Script running inside
> FreeSWITCH.
>
>
>
> The issue I have is fairly straightforward.
>
>
>
> I need a function to run when hangup is detected (ie. at the end of the
> call) however I understand this must not delay ending the script.  This
> function is CleanUp().  Then I would like the service to end.
>
>
>
> The problem I am having is if the caller hangs up during the playback of
> ?intro.wav? (as shown inside the MainService() function below), then the
> code jumps to the myHangupHook which calls CleanUp() perfectly, the issue
> is once CleanUp() is complete I would like the Lua script to end there and
> then (ie. at the bottom of CleanUp()).  What actually happens is it returns
> to MainService() and continues to try and play ?info.wav?, unless I either
> check for session:ready() everywhere or add a goto as shown below under
> each streamFile() function call.
>
>
>
> My aim is to reduce extra code and to make the Lua script simpler and
> easier to read.  Also I would like to try and avoid goto statements, which
> I know can be done with if (session:ready()) etc?.
>
>
>
> So is there a way to stop a Lua script running inside FreeSWITCH cleanly?
> I have tried the os.exit() this is barred from use by FreeSWITCH.  I have
> also tried session:destroy() which crashes FreeSWITCH (version 1.6.5 on
> CentOS 6.7, CentOS 7 and windows) 100% of the time!
>
>
>
> I could look further into the Lua additions done by the FreeSWITCH team in
> the source code, however if someone has already solved this then that would
> be the best solution.
>
>
>
> FYI: Obviously the script below is simple, however I am sure that you
> understand if the script was complicated having to use ?*if
> (session:ready()) then ?.?* or ?*if (not session:ready()) then goto
> HANGUPEXIT end?* makes the code ugly.
>
>
>
> Thanks in advance,
>
>
>
> Andrew Keil
>
> *Visytel Pty Ltd*
>
>
>
>
>
>
>
> ------------------------------------------------------------------------------
> Sample Lua Service
> -----------------------------------------------------------------------
>
>
>
> -- Lua template for FreeSWITCH service
>
> -- By: Andrew Keil (Visytel Pty Ltd)
>
> -- Email: support at visytel.com
>
>
>
> -- Setup script wide variables here
>
>
>
> function PreAnswer()
>
>                 freeswitch.consoleLog("INFO", "PRE ANSWER SECTION\n");
>
>                 -- Add your pre answer code from here
>
>
>
>                 -- End of your pre answer code
>
>                 freeswitch.consoleLog("INFO", "PRE ANSWER SECTION
> COMPLETE\n");
>
> end
>
>
>
> function AnswerCaller()
>
>                 session:answer()
>
>                 session:sleep(1000)
>
> end
>
>
>
> function MainService()
>
>                 freeswitch.consoleLog("INFO", "MAIN SERVICE
> SECTION\n");
>
>                 if (session:ready()) then
>
>                                 -- Note (1): If you wish to end the call
> then simply use: goto ENDSERVICE
>
>                                 -- Note (2): To terminate the service
> sooner when HANGUP is detected use: if (not session:ready()) then goto
> HANGUPEXIT end
>
>                                 -- Add your main service code from
> here               (caller would have been answered)
>
>
>
>                                 session:streamFile("intro.wav")
>
>                                 if (not session:ready()) then goto
> HANGUPEXIT end
>
>                                 session:streamFile("info.wav")
>
>                                 if (not session:ready()) then goto
> HANGUPEXIT end
>
>
> session:streamFile("outro.wav")
>
>                                 if (not session:ready()) then goto
> HANGUPEXIT end
>
>
>
>                                 -- End of your main service code
>
>                 end
>
>                 ::ENDSERVICE::
>
>                 if (session:ready()) then
>
>                                 -- End of service so hangup
>
>                                 session:hangup()  -- Should automatically
> jump to CleanUp() via hangup handler if caller still online at this stage
>
>                 end
>
>                 goto END
>
>                 ::HANGUPEXIT::
>
>                 freeswitch.consoleLog("INFO", "END OF SERVICE (HANGUP
> DETECTED)\n");
>
>                 ::END::
>
>                 freeswitch.consoleLog("INFO", "MAIN SERVICE SECTION
> COMPLETE\n");
>
> end
>
>
>
> function CleanUp()
>
>                 freeswitch.consoleLog("INFO", "CLEANUP SECTION\n");
>
>                 -- Add your cleanup code from here (caller would have been
> disconnected)
>
>
>
>                 -- End of your cleanup code
>
>                 freeswitch.consoleLog("INFO", "CLEANUP SECTION
> COMPLETE\n");
>
> end
>
>
>
> function myHangupHook(s, status, arg)
>
>                 session:hangup()
>
>                 CleanUp() -- Run CleanUp function now since the caller has
> disconnected
>
> end
>
>
>
> -- Setup Hangup event handler here
>
> v_hangup = "HANGUP"
>
> session:setHangupHook("myHangupHook", "v_hangup")
>
>
>
> -- Call service functions in order
>
> PreAnswer()
>
> AnswerCaller()
>
> MainService()
>
> -- End of Lua service
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From s.safarov at gmail.com  Fri Feb 26 17:44:53 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 26 Feb 2016 17:44:53 +0300
Subject: [Freeswitch-users] all-reg-options-ping and tls issue
In-Reply-To: <2E67ADAE1D3582409C90EBAE8C64C507122D35E9@QSCDEMXP01a.ONE4ALL.LAN>
References: <2E67ADAE1D3582409C90EBAE8C64C5070149F6B3@QSCDEMXP01a.ONE4ALL.LAN>
	<6D3DEA47-DECB-4509-A327-B8DFEC1C1A83@kavun.ch>
	<2E67ADAE1D3582409C90EBAE8C64C507122D35C1@QSCDEMXP01a.ONE4ALL.LAN>
	<CAHtxdDdy-nq_gZDLKW5bE62-5TQV2u-fK=sSmVD5Y07qv=0Cxw@mail.gmail.com>
	<2E67ADAE1D3582409C90EBAE8C64C507122D35E9@QSCDEMXP01a.ONE4ALL.LAN>
Message-ID: <CAHtxdDeAcZis=3SZnVwoK9j=29SLPTobDQUFGKSjLPfcFXGFTg@mail.gmail.com>

Please send output of command "sofia status profile internal reg
<exten_number>"

On Fri, Feb 26, 2016 at 4:21 PM, Seeliger, Adam <Adam.Seeliger at qsc.de>
wrote:

> Hi,
>
>
>
> the phone uses sip+tls.
>
> I test using a snom715, got plenty other phones here, but I guess they
> will behave the same way.
>
> It really looks like FreeSWITCH is doing something wrong (or is wrongly
> configured ? if there are any parameters for options ping supporting both,
> udp and tls)
>
>
>
> Regards
>
>
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Sergey
> Safarov
> *Gesendet:* Freitag, 26. Februar 2016 13:59
>
> *An:* FreeSWITCH Users Help
> *Betreff:* Re: [Freeswitch-users] all-reg-options-ping and tls issue
>
>
>
> If your phone has enabled SIPS uri please disable and use sip+tls.
>
>
>
>
>
> On Fri, Feb 26, 2016 at 3:37 PM, Seeliger, Adam <Adam.Seeliger at qsc.de>
> wrote:
>
> Hi and thanks for the feedback,
>
>
>
> sry that I did not respond for a long time.
>
>
>
> I already use:
>
> <variable name=?sip-force-contact?
> value="NDLB-connectile-dysfunction-2.0"/>
>
>
>
> I also tested all mentioned params below, nothing works.
>
>
>
> When I register a User via TLS FreeSWITCH does not even try to ping the
> user.
>
> I turned sofia global siptrace on and watched the flow:
>
>
>
> User                                        Server
>
> 13:09:33.311446:             REGISTER [TLS] ->
>
> 13:09:33.312552:             <- 401 UNAUTHORIZED [TLS]
>
> 13:09:33.331948:             REGISTER (AUTH) [TLS]  ->
>
> 13:09:33.336959:             <- 200 OK [TLS]
>
> Nothing happens
>
> 2016-02-26 13:10:00.619525 [WARNING] sofia.c:5769 Sip user 'user at host' is
> now Unreachable
>
> 2016-02-26 13:10:00.619525 [WARNING] sofia.c:5780 Expire sip user
> 'user at host' due to options failure
>
>
>
> When I REGISTER the User via UDP FreeSWITCH starts to ping (OPTIONS) the
> user as soon as he is registered.
>
>
>
> Is there any way to force FreeSWITCH to send OPTIONs in both, udp and tls,
> depending on the registration?
>
>
>
> Thanks in advance,
>
> Adam
>
>
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Emrah
> *Gesendet:* Freitag, 29. Januar 2016 09:25
> *An:* FreeSWITCH Users Help
> *Betreff:* Re: [Freeswitch-users] all-reg-options-ping and tls issue
>
>
>
> Hi!
>
> This is interesting. I experienced something rather similar where calls
> would drop because FS would timeout on certain packets sent over UDP
> instead of TLS.
>
> I assume you mean FS exits with port 5060 instead of port 5061? Because
> the port on the remote end should be dynamically set.
>
> I found out that in my case, what works best even with TLS, is to use:
>
> <variable
> name=?sip-force-contact? value="NDLB-connectile-dysfunction-2.0"/>
>
> This goes as far as it can to lay out the path to contacting the client
> with all consideration in regards to NAT and dynamic ports.
>
> Not sure if it will help you. I?ve personally disabled options-ping an let
> my clients deal with keep-alive instead.
>
>
>
> You could also look into:
>
>     <!-- add a ;received="<ip>:<port>" to the contact when replying to
> register for nat handling -->
>
> <!? <param name="NDLB-received-in-nat-reg-contact" value="true"/> -->
>
> <!? <param name="bind-params" value="transport=tcp,udp,tls"/>  -->
>
> <!-- additional bind parameters for TLS -->
>
>     <param name=?tls-bind-params" value="transport=tls"/>
>
>
>
> I?ll leave it up to you to investigate those options more in details on
> the FS documentation.
>
>
>
> Please keep us posted!
>
>
>
> E
>
> On Jan 28, 2016, at 11:48 AM, Seeliger, Adam <Adam.Seeliger at qsc.de> wrote:
>
>
>
> Hi all,
>
>
>
> I have a problem, when I enable TLS and register a phone using TLS on Port
> 5061.
>
> FreeSWITCH still tries to ?ping? the phone using Port 5060 using UDP,
> which is ignored by the phone.
>
> Moments later FreeSWITCH deletes the registration, because
> ?unregister-on-options-fail? is set to ?true?.
>
>
>
> I already figured out, that you can set ?all-reg-options-ping? to
> ?udp-only?, but this would completely disable this feature for TLS.
>
> Is there any way to ping TLS registered using TLS?
>
>
>
> Thanks in advance
>
>
>
> - Adam
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Adam.Seeliger at qsc.de  Fri Feb 26 17:59:14 2016
From: Adam.Seeliger at qsc.de (Seeliger, Adam)
Date: Fri, 26 Feb 2016 14:59:14 +0000
Subject: [Freeswitch-users] all-reg-options-ping and tls issue
In-Reply-To: <CAHtxdDeAcZis=3SZnVwoK9j=29SLPTobDQUFGKSjLPfcFXGFTg@mail.gmail.com>
References: <2E67ADAE1D3582409C90EBAE8C64C5070149F6B3@QSCDEMXP01a.ONE4ALL.LAN>
	<6D3DEA47-DECB-4509-A327-B8DFEC1C1A83@kavun.ch>
	<2E67ADAE1D3582409C90EBAE8C64C507122D35C1@QSCDEMXP01a.ONE4ALL.LAN>
	<CAHtxdDdy-nq_gZDLKW5bE62-5TQV2u-fK=sSmVD5Y07qv=0Cxw@mail.gmail.com>
	<2E67ADAE1D3582409C90EBAE8C64C507122D35E9@QSCDEMXP01a.ONE4ALL.LAN>
	<CAHtxdDeAcZis=3SZnVwoK9j=29SLPTobDQUFGKSjLPfcFXGFTg@mail.gmail.com>
Message-ID: <2E67ADAE1D3582409C90EBAE8C64C507122D361C@QSCDEMXP01a.ONE4ALL.LAN>

Hi, here is the requested output.

I changed the real user, domain and ip address values into descriptions.
I guess the values are not necessary?

Both, FreeSWITCH and the phone are in the same network (no NAT involved here)

Registrations:
=================================================================================================
Call-ID:         3134353634383833373032323631-ncwgvit2obfp
User:            user at domain
Contact:       "User Name" <sip:user at ip:60206;transport=tls;line=gvg9q8jh;fs_nat=yes;fs_path=sip%3Auser%40ip%3A60206>
Agent:          snom715/8.7.5.35
Status:         Registered(AUTO-NAT-2.0)(unknown) EXP(2016-02-26 16:21:32) EXPSECS(1912)
Ping-Status:   Reachable
Host:            hostname
IP:               ip
Port:            60206
Auth-User:    user
Auth-Realm: domain
MWI-Account:          user at domain

Total items returned: 1
=================================================================================================
2016-02-26 15:50:09.326648 [WARNING] sofia.c:5769 Sip user 'user at domain' is now Unreachable
2016-02-26 15:50:09.326648 [WARNING] sofia.c:5780 Expire sip user 'user at domain' due to options failure

As you can see, the user immediately got unregistered again ?

Best regards,
Adam

Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Sergey Safarov
Gesendet: Freitag, 26. Februar 2016 15:45
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] all-reg-options-ping and tls issue

Please send output of command "sofia status profile internal reg <exten_number>"

On Fri, Feb 26, 2016 at 4:21 PM, Seeliger, Adam <Adam.Seeliger at qsc.de<mailto:Adam.Seeliger at qsc.de>> wrote:
Hi,

the phone uses sip+tls.
I test using a snom715, got plenty other phones here, but I guess they will behave the same way.
It really looks like FreeSWITCH is doing something wrong (or is wrongly configured ? if there are any parameters for options ping supporting both, udp and tls)

Regards

Von: 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>] Im Auftrag von Sergey Safarov
Gesendet: Freitag, 26. Februar 2016 13:59

An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] all-reg-options-ping and tls issue

If your phone has enabled SIPS uri please disable and use sip+tls.


On Fri, Feb 26, 2016 at 3:37 PM, Seeliger, Adam <Adam.Seeliger at qsc.de<mailto:Adam.Seeliger at qsc.de>> wrote:
Hi and thanks for the feedback,

sry that I did not respond for a long time.

I already use:
<variable name=?sip-force-contact? value="NDLB-connectile-dysfunction-2.0"/>

I also tested all mentioned params below, nothing works.

When I register a User via TLS FreeSWITCH does not even try to ping the user.
I turned sofia global siptrace on and watched the flow:

User                                        Server
13:09:33.311446:             REGISTER [TLS] ->
13:09:33.312552:             <- 401 UNAUTHORIZED [TLS]
13:09:33.331948:             REGISTER (AUTH) [TLS]  ->
13:09:33.336959:             <- 200 OK [TLS]
Nothing happens
2016-02-26 13:10:00.619525 [WARNING] sofia.c:5769 Sip user 'user at host' is now Unreachable
2016-02-26 13:10:00.619525 [WARNING] sofia.c:5780 Expire sip user 'user at host' due to options failure

When I REGISTER the User via UDP FreeSWITCH starts to ping (OPTIONS) the user as soon as he is registered.

Is there any way to force FreeSWITCH to send OPTIONs in both, udp and tls, depending on the registration?

Thanks in advance,
Adam

Von: 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>] Im Auftrag von Emrah
Gesendet: Freitag, 29. Januar 2016 09:25
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] all-reg-options-ping and tls issue

Hi!
This is interesting. I experienced something rather similar where calls would drop because FS would timeout on certain packets sent over UDP instead of TLS.
I assume you mean FS exits with port 5060 instead of port 5061? Because the port on the remote end should be dynamically set.
I found out that in my case, what works best even with TLS, is to use:
<variable name=?sip-force-contact? value="NDLB-connectile-dysfunction-2.0"/>
This goes as far as it can to lay out the path to contacting the client with all consideration in regards to NAT and dynamic ports.
Not sure if it will help you. I?ve personally disabled options-ping an let my clients deal with keep-alive instead.

You could also look into:
    <!-- add a ;received="<ip>:<port>" to the contact when replying to register for nat handling -->
<!? <param name="NDLB-received-in-nat-reg-contact" value="true"/> -->
<!? <param name="bind-params" value="transport=tcp,udp,tls"/>  -->
<!-- additional bind parameters for TLS -->
    <param name=?tls-bind-params" value="transport=tls"/>

I?ll leave it up to you to investigate those options more in details on the FS documentation.

Please keep us posted!

E
On Jan 28, 2016, at 11:48 AM, Seeliger, Adam <Adam.Seeliger at qsc.de<mailto:Adam.Seeliger at qsc.de>> wrote:

Hi all,

I have a problem, when I enable TLS and register a phone using TLS on Port 5061.
FreeSWITCH still tries to ?ping? the phone using Port 5060 using UDP, which is ignored by the phone.
Moments later FreeSWITCH deletes the registration, because ?unregister-on-options-fail? is set to ?true?.

I already figured out, that you can set ?all-reg-options-ping? to ?udp-only?, but this would completely disable this feature for TLS.
Is there any way to ping TLS registered using TLS?

Thanks in advance

- Adam

_________________________________________________________________________
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 brian at freeswitch.org  Fri Feb 26 18:15:04 2016
From: brian at freeswitch.org (Brian West)
Date: Fri, 26 Feb 2016 09:15:04 -0600
Subject: [Freeswitch-users] Call gets disconnected on write frame error
In-Reply-To: <a5dbdc1387a84d32b76b6a42d1eb1be9@UK-MAIL-001.edge.local>
References: <a5dbdc1387a84d32b76b6a42d1eb1be9@UK-MAIL-001.edge.local>
Message-ID: <CAEJMVkCHv14k0OT2GEmaYJQVrcuSWXwm-A8Q1UgqiLe31GhUJA@mail.gmail.com>

What revision of FreeSWITCH are you running?

On Fri, Feb 26, 2016 at 8:20 AM, Chris Young <Chris.Young at enghouse.com>
wrote:

> Hi all,
>
>
>
> Recently, we've been experiencing a problem whereby FreeSWITCH disconnects
> a call after receiving a frame of audio data with a data length of 0. When
> this happens we see the following messages in the log:
>
>
>
>                 [DEBUG] switch_core_io.c:1503 Engaging Write Buffer at 320
> bytes to accommodate 0->320
>
>                 [ERR] switch_core_io.c:1518 Write Buffer 0 bytes Failed!
>
>
>
> and the call gets disconnected. This normally happens when trying to join
> a conference but we have also seen it when bridging two calls together.
>
>
>
> Is the call disconnection by-design or just a side-effect of having
> received the bad frame of data?
>
>
>
> Thanks,
>
> Chris
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

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 Chris.Young at enghouse.com  Fri Feb 26 18:25:25 2016
From: Chris.Young at enghouse.com (Chris Young)
Date: Fri, 26 Feb 2016 15:25:25 +0000
Subject: [Freeswitch-users] Call gets disconnected on write frame error
In-Reply-To: <CAEJMVkCHv14k0OT2GEmaYJQVrcuSWXwm-A8Q1UgqiLe31GhUJA@mail.gmail.com>
References: <a5dbdc1387a84d32b76b6a42d1eb1be9@UK-MAIL-001.edge.local>
	<CAEJMVkCHv14k0OT2GEmaYJQVrcuSWXwm-A8Q1UgqiLe31GhUJA@mail.gmail.com>
Message-ID: <4a445ea4ec26444a986cf29ba2bc20c4@UK-MAIL-001.edge.local>

1.4.18.

I realise that this is not the most up-to-date version but I'm not in a position to re-test with a newer release at the moment.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: 26 February 2016 15:15
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Call gets disconnected on write frame error

What revision of FreeSWITCH are you running?

On Fri, Feb 26, 2016 at 8:20 AM, Chris Young <Chris.Young at enghouse.com<mailto:Chris.Young at enghouse.com>> wrote:
Hi all,

Recently, we've been experiencing a problem whereby FreeSWITCH disconnects a call after receiving a frame of audio data with a data length of 0. When this happens we see the following messages in the log:

                [DEBUG] switch_core_io.c:1503 Engaging Write Buffer at 320 bytes to accommodate 0->320
                [ERR] switch_core_io.c:1518 Write Buffer 0 bytes Failed!

and the call gets disconnected. This normally happens when trying to join a conference but we have also seen it when bridging two calls together.

Is the call disconnection by-design or just a side-effect of having received the bad frame of data?

Thanks,
Chris

_________________________________________________________________________
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



--

Brian West
brian at freeswitch.org<mailto:brian at freeswitch.org>

[http://billing.freeswitch.org/templates/default/img/whmcslogo.png]

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

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 s.safarov at gmail.com  Fri Feb 26 18:52:30 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 26 Feb 2016 15:52:30 +0000
Subject: [Freeswitch-users] all-reg-options-ping and tls issue
In-Reply-To: <2E67ADAE1D3582409C90EBAE8C64C507122D361C@QSCDEMXP01a.ONE4ALL.LAN>
References: <2E67ADAE1D3582409C90EBAE8C64C5070149F6B3@QSCDEMXP01a.ONE4ALL.LAN>
	<6D3DEA47-DECB-4509-A327-B8DFEC1C1A83@kavun.ch>
	<2E67ADAE1D3582409C90EBAE8C64C507122D35C1@QSCDEMXP01a.ONE4ALL.LAN>
	<CAHtxdDdy-nq_gZDLKW5bE62-5TQV2u-fK=sSmVD5Y07qv=0Cxw@mail.gmail.com>
	<2E67ADAE1D3582409C90EBAE8C64C507122D35E9@QSCDEMXP01a.ONE4ALL.LAN>
	<CAHtxdDeAcZis=3SZnVwoK9j=29SLPTobDQUFGKSjLPfcFXGFTg@mail.gmail.com>
	<2E67ADAE1D3582409C90EBAE8C64C507122D361C@QSCDEMXP01a.ONE4ALL.LAN>
Message-ID: <CAHtxdDfuv86b8a0EsG=6yW3QD1KwXbmtNs5_7O6SnpbX1PFJ=g@mail.gmail.com>

Registration is correct.
Think it bug and requred to fill a jira tiket.
I use master 4 mouth old and it work correctly.

On Fri, Feb 26, 2016, 18:00 Seeliger, Adam <Adam.Seeliger at qsc.de> wrote:

> Hi, here is the requested output.
>
>
>
> I changed the real user, domain and ip address values into descriptions.
>
> I guess the values are not necessary?
>
>
>
> Both, FreeSWITCH and the phone are in the same network (no NAT involved
> here)
>
>
>
> Registrations:
>
>
> =================================================================================================
>
> Call-ID:         3134353634383833373032323631-ncwgvit2obfp
>
> User:            user at domain
>
> Contact:       "User Name" <sip:user at ip
> :60206;transport=tls;line=gvg9q8jh;fs_nat=yes;fs_path=sip%3Auser%40ip%3A60206>
>
> Agent:          snom715/8.7.5.35
>
> Status:         Registered(AUTO-NAT-2.0)(unknown) EXP(2016-02-26 16:21:32)
> EXPSECS(1912)
>
> Ping-Status:   Reachable
>
> Host:            hostname
>
> IP:               ip
>
> Port:            60206
>
> Auth-User:    user
>
> Auth-Realm: domain
>
> MWI-Account:          user at domain
>
>
>
> Total items returned: 1
>
>
> =================================================================================================
>
> 2016-02-26 15:50:09.326648 [WARNING] sofia.c:5769 Sip user 'user at domain'
> is now Unreachable
>
> 2016-02-26 15:50:09.326648 [WARNING] sofia.c:5780 Expire sip user
> 'user at domain' due to options failure
>
>
>
> As you can see, the user immediately got unregistered again L
>
>
>
> Best regards,
>
> Adam
>
>
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Sergey
> Safarov
> *Gesendet:* Freitag, 26. Februar 2016 15:45
>
>
> *An:* FreeSWITCH Users Help
> *Betreff:* Re: [Freeswitch-users] all-reg-options-ping and tls issue
>
>
>
> Please send output of command "sofia status profile internal reg
> <exten_number>"
>
>
>
> On Fri, Feb 26, 2016 at 4:21 PM, Seeliger, Adam <Adam.Seeliger at qsc.de>
> wrote:
>
> Hi,
>
>
>
> the phone uses sip+tls.
>
> I test using a snom715, got plenty other phones here, but I guess they
> will behave the same way.
>
> It really looks like FreeSWITCH is doing something wrong (or is wrongly
> configured ? if there are any parameters for options ping supporting both,
> udp and tls)
>
>
>
> Regards
>
>
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Sergey
> Safarov
> *Gesendet:* Freitag, 26. Februar 2016 13:59
>
>
> *An:* FreeSWITCH Users Help
> *Betreff:* Re: [Freeswitch-users] all-reg-options-ping and tls issue
>
>
>
> If your phone has enabled SIPS uri please disable and use sip+tls.
>
>
>
>
>
> On Fri, Feb 26, 2016 at 3:37 PM, Seeliger, Adam <Adam.Seeliger at qsc.de>
> wrote:
>
> Hi and thanks for the feedback,
>
>
>
> sry that I did not respond for a long time.
>
>
>
> I already use:
>
> <variable name=?sip-force-contact?
> value="NDLB-connectile-dysfunction-2.0"/>
>
>
>
> I also tested all mentioned params below, nothing works.
>
>
>
> When I register a User via TLS FreeSWITCH does not even try to ping the
> user.
>
> I turned sofia global siptrace on and watched the flow:
>
>
>
> User                                        Server
>
> 13:09:33.311446:             REGISTER [TLS] ->
>
> 13:09:33.312552:             <- 401 UNAUTHORIZED [TLS]
>
> 13:09:33.331948:             REGISTER (AUTH) [TLS]  ->
>
> 13:09:33.336959:             <- 200 OK [TLS]
>
> Nothing happens
>
> 2016-02-26 13:10:00.619525 [WARNING] sofia.c:5769 Sip user 'user at host' is
> now Unreachable
>
> 2016-02-26 13:10:00.619525 [WARNING] sofia.c:5780 Expire sip user
> 'user at host' due to options failure
>
>
>
> When I REGISTER the User via UDP FreeSWITCH starts to ping (OPTIONS) the
> user as soon as he is registered.
>
>
>
> Is there any way to force FreeSWITCH to send OPTIONs in both, udp and tls,
> depending on the registration?
>
>
>
> Thanks in advance,
>
> Adam
>
>
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Emrah
> *Gesendet:* Freitag, 29. Januar 2016 09:25
> *An:* FreeSWITCH Users Help
> *Betreff:* Re: [Freeswitch-users] all-reg-options-ping and tls issue
>
>
>
> Hi!
>
> This is interesting. I experienced something rather similar where calls
> would drop because FS would timeout on certain packets sent over UDP
> instead of TLS.
>
> I assume you mean FS exits with port 5060 instead of port 5061? Because
> the port on the remote end should be dynamically set.
>
> I found out that in my case, what works best even with TLS, is to use:
>
> <variable
> name=?sip-force-contact? value="NDLB-connectile-dysfunction-2.0"/>
>
> This goes as far as it can to lay out the path to contacting the client
> with all consideration in regards to NAT and dynamic ports.
>
> Not sure if it will help you. I?ve personally disabled options-ping an let
> my clients deal with keep-alive instead.
>
>
>
> You could also look into:
>
>     <!-- add a ;received="<ip>:<port>" to the contact when replying to
> register for nat handling -->
>
> <!? <param name="NDLB-received-in-nat-reg-contact" value="true"/> -->
>
> <!? <param name="bind-params" value="transport=tcp,udp,tls"/>  -->
>
> <!-- additional bind parameters for TLS -->
>
>     <param name=?tls-bind-params" value="transport=tls"/>
>
>
>
> I?ll leave it up to you to investigate those options more in details on
> the FS documentation.
>
>
>
> Please keep us posted!
>
>
>
> E
>
> On Jan 28, 2016, at 11:48 AM, Seeliger, Adam <Adam.Seeliger at qsc.de> wrote:
>
>
>
> Hi all,
>
>
>
> I have a problem, when I enable TLS and register a phone using TLS on Port
> 5061.
>
> FreeSWITCH still tries to ?ping? the phone using Port 5060 using UDP,
> which is ignored by the phone.
>
> Moments later FreeSWITCH deletes the registration, because
> ?unregister-on-options-fail? is set to ?true?.
>
>
>
> I already figured out, that you can set ?all-reg-options-ping? to
> ?udp-only?, but this would completely disable this feature for TLS.
>
> Is there any way to ping TLS registered using TLS?
>
>
>
> Thanks in advance
>
>
>
> - Adam
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Fri Feb 26 20:38:34 2016
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Fri, 26 Feb 2016 09:38:34 -0800
Subject: [Freeswitch-users] Transcoding and recording
Message-ID: <CAC8JMOJ4rGtOKCDpKOKXTugXXD3S_TUXEuruJCLMjiZQipGZAA@mail.gmail.com>

Hi guys,

if I record a session where both A and B use the same codec, will
transcoding occur?

Basically, in this case will the same codecs gain any CPU advantage?
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From brian at freeswitch.org  Fri Feb 26 21:02:17 2016
From: brian at freeswitch.org (Brian West)
Date: Fri, 26 Feb 2016 12:02:17 -0600
Subject: [Freeswitch-users] Transcoding and recording
In-Reply-To: <CAC8JMOJ4rGtOKCDpKOKXTugXXD3S_TUXEuruJCLMjiZQipGZAA@mail.gmail.com>
References: <CAC8JMOJ4rGtOKCDpKOKXTugXXD3S_TUXEuruJCLMjiZQipGZAA@mail.gmail.com>
Message-ID: <CAEJMVkA7DwVpi+-dPzB0q_h4CqsxUvdFAM8X5K_hifMmHzSZJg@mail.gmail.com>

Yes, transcoding is required to record a mux'ed stream or a stereo stream
into a wav file.

On Fri, Feb 26, 2016 at 11:38 AM, Oleg Stolyar <olegstolyar at gmail.com>
wrote:

> Hi guys,
>
> if I record a session where both A and B use the same codec, will
> transcoding occur?
>
> Basically, in this case will the same codecs gain any CPU advantage?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

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 olegstolyar at gmail.com  Fri Feb 26 21:21:21 2016
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Fri, 26 Feb 2016 10:21:21 -0800
Subject: [Freeswitch-users] Transcoding and recording
In-Reply-To: <CAEJMVkA7DwVpi+-dPzB0q_h4CqsxUvdFAM8X5K_hifMmHzSZJg@mail.gmail.com>
References: <CAC8JMOJ4rGtOKCDpKOKXTugXXD3S_TUXEuruJCLMjiZQipGZAA@mail.gmail.com>
	<CAEJMVkA7DwVpi+-dPzB0q_h4CqsxUvdFAM8X5K_hifMmHzSZJg@mail.gmail.com>
Message-ID: <CAC8JMOLyPb5aaij5TrjOCDJoZqnJq0v9ytG_mc4HewnnBiBXUQ@mail.gmail.com>

Thanks Brian!

On Fri, Feb 26, 2016 at 10:02 AM, Brian West <brian at freeswitch.org> wrote:

> Yes, transcoding is required to record a mux'ed stream or a stereo stream
> into a wav file.
>
> On Fri, Feb 26, 2016 at 11:38 AM, Oleg Stolyar <olegstolyar at gmail.com>
> wrote:
>
>> Hi guys,
>>
>> if I record a session where both A and B use the same codec, will
>> transcoding occur?
>>
>> Basically, in this case will the same codecs gain any CPU advantage?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> 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 blasterjr at gmail.com  Fri Feb 26 21:28:47 2016
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Fri, 26 Feb 2016 11:28:47 -0700
Subject: [Freeswitch-users] How to kill call/channel on the reception of
 CUSTOM avmd::beep
In-Reply-To: <CAGVOjVUzZQ2sVOejJyErMNJTRFOMrzeb+v+1gasDZNM7DUFvWQ@mail.gmail.com>
References: <CAGVOjVUT0L8uZ_Fspv9CAnq1Vnn1tXyMKnLmmfSYW5qDD+-zKQ@mail.gmail.com>
	<CAO3hyzG_xaLhUHT-qEgB2nxQYsAWjNO0QCGdWdA=Kfxf0zWL3g@mail.gmail.com>
	<CAGVOjVXa723vJc+hq-fYzA0X0yvF7im+Bu4u18DJHSa-YS4ByA@mail.gmail.com>
	<CAGVOjVUzZQ2sVOejJyErMNJTRFOMrzeb+v+1gasDZNM7DUFvWQ@mail.gmail.com>
Message-ID: <CAO3hyzHrfjL5c4Vdew1O3wDsc83wUV79TOO5o69_K9N8b=SiRQ@mail.gmail.com>

Glad to hear and i'm glad i could help.

On Thu, Feb 25, 2016 at 2:15 PM, Aqs Younas <aqsyounas at gmail.com> wrote:

> My fault. Got it working with uuid_kill Unique-ID
>
> Thanks.
>
> On 26 February 2016 at 01:01, Aqs Younas <aqsyounas at gmail.com> wrote:
>
>> Thanks for your Reply.
>>
>> freeswitch at debian> uuid_hangup Unique-ID
>> Unknown Command: uuid_hangup 11e64e66-f81b-4d23-a8e1-e6af0eaa0bbc
>>
>> Tried to kill.
>>
>> freeswitch at debian> uuid_kill Unique-ID
>>
>> -ERR No such channel!
>>
>> Any suggestion?
>>
>> On 25 February 2016 at 23:43, Chris Tunbridge <blasterjr at gmail.com>
>> wrote:
>>
>>> uuid_hangup the value of the Unique-ID field possibly?
>>>
>>> On Thu, Feb 25, 2016 at 10:13 AM, Aqs Younas <aqsyounas at gmail.com>
>>> wrote:
>>>
>>>> Hi,
>>>>
>>>> I have subscribed to CUSTOM avmd::beep and this is what I get.
>>>>
>>>>         "Event-Subclass":       "avmd::beep",
>>>>         "Event-Name":   "CUSTOM",
>>>>         "Core-UUID":    "11e64e66-f81b-4d23-a8e1-e6af0eaa0bbc",
>>>>         "FreeSWITCH-Hostname":  "debian",
>>>>         "FreeSWITCH-Switchname":        "debian",
>>>>         "FreeSWITCH-IPv4":      "192.168.10.59",
>>>>         "FreeSWITCH-IPv6":      "::1",
>>>>         "Event-Date-Local":     "2016-02-25 10:46:17",
>>>>         "Event-Date-GMT":       "Thu, 25 Feb 2016 15:46:17 GMT",
>>>>         "Event-Date-Timestamp": "1456415177487145",
>>>>         "Event-Calling-File":   "mod_avmd.c",
>>>>         "Event-Calling-Function":       "avmd_process",
>>>>         "Event-Calling-Line-Number":    "550",
>>>>         "Event-Sequence":       "3174",
>>>>         "Beep-Status":  "stop",
>>>>         "Unique-ID":    "80d33754-b234-47e2-b7c5-16ffa20f359c",
>>>>         "call-command": "avmd"
>>>>
>>>> Now I need to hangup that call(beep detected), but could not find any
>>>> event variable connected to the call.
>>>>
>>>> How can i kill the call.
>>>>
>>>> Any pointer is much appreciated.
>>>>
>>>> This email has been sent from a virus-free computer protected by Avast.
>>>> www.avast.com
>>>> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
>>>> <#-2020178404_996209302_-2019354057_-1458571440_DDB4FAA8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mario_fs at mgtech.com  Fri Feb 26 22:21:57 2016
From: mario_fs at mgtech.com (Mario G)
Date: Fri, 26 Feb 2016 11:21:57 -0800
Subject: [Freeswitch-users] major build change in master.
In-Reply-To: <8D88662E-BB53-49F2-B003-2C9288A3FD03@jerris.com>
References: <F7862F99-22B2-4165-8EE3-9872E848B14E@jerris.com>
	<8C3827C6-22F2-45BB-BFAA-6AB64BD67E12@mgtech.com>
	<01e201d16ffb$db648f10$922dad30$@botecomm.com>
	<8D88662E-BB53-49F2-B003-2C9288A3FD03@jerris.com>
Message-ID: <4D490B11-E40A-4598-B480-F203F1DAAF46@mgtech.com>

FYI in case this affects Linux/Windows: removal of libvpx prereq install requires nasm or yasm to be added as prerequisite under OS X 10.9/10/11 since the OS X nasm is too old. Previous hombrew libvpx auto installed yasm.

> On Feb 25, 2016, at 10:49 AM, Michael Jerris <mike at jerris.com> wrote:
> 
> we are working on how exactly this will effect packaging and install instructions, but we will no longer have yuv or vpx pre-reqs and the requirement to rebuild libav against our vpx is also removed.
> 
>> On Feb 25, 2016, at 1:39 PM, Bote Man <bote_radio at botecomm.com> wrote:
>> 
>> Mike, I put a note to this effect on the Debian 8 installation page.
>> 
>> I'm glad I caught this as I have been on a customer's site a lot and have
>> not been able to keep with the -users mailing list. Lemme know if this
>> changes so I can update it.
>> 
>> Thanks.
>> 
>> Bote
>> 
>> 
>>> -----Original Message-----
>>> From: Mario G
>>> Sent: Thursday, 25 February, 2016 13:16
>>> To: FreeSWITCH Users Help
>>> Subject: Re: [Freeswitch-users] major build change in master.
>>> 
>>> This fixed OS X VPX issue, aok now. Questions:
>>> 
>>> I assume this means YUV and VPX can be removed as prerequisites?
>>> 
>>> Will this be one for 1.6 as well?
>>> 
>>> Just asking so I can update the wiki and Applescript installer as needed.
>>> Thanks!
>>> 
>>> 
>>>> On Feb 24, 2016, at 3:49 PM, Michael Jerris <mike at jerris.com> wrote:
>>>> 
>>>> please note this important change that went into master today.  As of
>>> today, we are no longer using system versions of libyuv and libvpx due to
>>> major conflicts with system versions of these libraries.  These are now
>> built
>>> static into the freeswitch core.  Also note, mod_vpx no longer exists, it
>> is
>>> automatically loaded as part of the core and you will no longer have
>>> mod_vpx.so or have to manually load it.  I'll have more details coming,
>> but let
>>> me know if you have any questions.  Please note for anyone doing cross
>>> compiling, this probably needs a bit more work for you, I'll be working on
>>> fixing that this week.
>>>> 
>>>> Mike
>>>> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 freeswitch at digitaldescent.net  Sat Feb 27 01:45:03 2016
From: freeswitch at digitaldescent.net (Brian Chow)
Date: Fri, 26 Feb 2016 14:45:03 -0800
Subject: [Freeswitch-users] Inbound calls mis-routing routing to
 internal extension with external IP?
In-Reply-To: <62e0a3e6c4e14d40b15aedb1d3e1b3b6@nysolutions.com>
References: <CAA_Md=uAGmnqLkY3g128QcjoBis4VnoTwSUEZpqAws35AwRsHQ@mail.gmail.com>
	<23b1626b3814448b90842a279dc30be5@nysolutions.com>
	<CAA_Md=t4NR-ocAsnvOmD39XGNAexYVZWsWy4taYbiXU7-N1qEg@mail.gmail.com>
	<CAEJMVkC+=U50t2nEYrn9CU1OQnXOUcVXKmpzaB8KF=Gbh7T-BQ@mail.gmail.com>
	<CAA_Md=v=PGrR_0s90SK_ijZO5eScqJYN+Utqu22nwqbZEQXvLA@mail.gmail.com>
	<CAA_Md=sTVGHHmAvtV9EV2OvBuLkdnQ3b9wRaSCMYjBwf71p-yA@mail.gmail.com>
	<bf0ede9ef70640818f0607ceb46f5434@nysolutions.com>
	<CAA_Md=tJGhw7_BGz4qFhXAz_dnpy8d7+2L0cBRm6ZaF9jPeHaQ@mail.gmail.com>
	<62e0a3e6c4e14d40b15aedb1d3e1b3b6@nysolutions.com>
Message-ID: <CAA_Md=v943_=QbEWo4BxPK7g4UNnPkFN97GaEn6mwZie6aeoAA@mail.gmail.com>

I think that solved it, thanks!  I did have ACL configured, just apparently
incorrectly.  Out of the (fusionpbx) box I had two ACL lists, one called
LAN with a default of allow, which I had entered my CIDR into.  One called
domain, which was default deny, but I had made an allow entry for my
domain, but no CIDR.  When I added my CIDR to the domain list, it started
working.

Thanks again for walking me through the troubleshooting, sorry for such a
noob mistake :)

On Thu, Feb 25, 2016 at 5:50 PM, Moishe Grunstein <max at nysolutions.com>
wrote:

> Can you show a log of a internal call?
>
> Are the phones showing registered? Show registrations
>
> Did you by any chance add your ip range to the ACL.
>
>
>
>
>
> 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 *Brian Chow
> *Sent:* Thursday, February 25, 2016 8:18 PM
>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Inbound calls mis-routing routing to
> internal extension with external IP?
>
>
>
> Moishe,
>
>   Thanks for double checking this, I hadn't thought to try an extension
> <-> extension call as it's a private server just for me, however, after
> creating another extension, I cannot make internal calls.  It seems, I can
> only make outbound calls.
>
>
>
> Internal -> External is working.
>
> External -> Internal only works to the FS server, not to the extension.
>
> Internal -> Internal does not work.
>
>
>
>
>
>
>
> On Thu, Feb 25, 2016 at 8:12 AM, Moishe Grunstein <max at nysolutions.com>
> wrote:
>
> Does internal call also go direct to voicemail?
>
> Are you sure you don?t have a *99 before the extension number?
>
>
>
> 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 *Brian Chow
> *Sent:* Thursday, February 25, 2016 11:00 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Inbound calls mis-routing routing to
> internal extension with external IP?
>
>
>
> Ok, in that case, what do I do next to figure out why inbound calls go
> directly to voicemail instead of ringing the extension?
>
> On Feb 25, 2016 7:57 AM, "Brian Chow" <brianchow at digitaldescent.net>
> wrote:
>
> Ok, in that case, what do I look at next to see why it goes straight to
> voicemail instead of ringing my extension?
>
> On Feb 25, 2016 5:52 AM, "Brian West" <brian at freeswitch.org> wrote:
>
> Its just the channel name so don't get tripped up by that, its just named
> with some default data about the session.
>
>
>
> On Wed, Feb 24, 2016 at 11:33 PM, Brian Chow <
> freeswitch at digitaldescent.net> wrote:
>
> Debug console output: https://pastebin.freeswitch.org/24572
>
>
>
> My fqdn only resolves inside the network, but the external-ip is
> configured in FS to use stun, which reports my external ip.
>
>
>
> >Is your endpoint registered? How often is it set to reregister?
>
> Yes, and the zoiper default registration expiry is 3600
>
>
>
> >Does your router have a sip alg? Are the ports opened in your firewall?
>
> > ish, no GUI for it, but I did make sure to disable it from the command
> line (Ubiquiti Edgerouter).  No firewall ports are explicitly open, but
> allow bi-directional with existing states.  I do get 2 way audio when
> calling out, and I also can hear the system forwarding my call in to
> voicemail; the voicemail message and prompts.  I know asterisk is different
> from FS, but my asterisk experience has proven that at least with asterisk,
> port forwarding isn't needed.  Also, unfortunately, I cannot get a static
> ip.
>
>
>
> Thanks!
>
>   -Brian
>
>
>
> On Wed, Feb 24, 2016 at 8:31 PM, Moishe Grunstein <max at nysolutions.com>
> wrote:
>
> It is probably sending call to your extension at domainname, if your
> external ip is your domain name then you will see the external ip.
>
> Is your endpoint registered? How often is it set to reregister?
>
> Does your router have a sip alg? Are the ports opened in your firewall?
>
> Very hard to guess without a log of a call with debug enabled.
>
> sofia global sip trace on
>
>
>
> 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 *Brian Chow
> *Sent:* Wednesday, February 24, 2016 10:00 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Inbound calls mis-routing routing to
> internal extension with external IP?
>
>
>
> Hello all!
>
>   I'm new to freeswitch, so I'm sure this is just a newbie configuration
> error.  Sorry if it's been answered a million times, my google searches
> always just bring up the standard NAT configuration pages.  I've already
> followed the confluence page on configuring NAT.
>
>
>
> My Setup:
>
> Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3
> instance.
>
> SIP Provider: Flowroute
>
>
>
> NAT: extension and freeswitch are on the same network, both of which are
> behind NAT and connecting to flowroute.  I configured external sip and rtp
> to use stun entries.  External profile is using $${external_rtp/sip_ip} for
> ext-rtp/sip respectively.
>
>
>
> I have one DID configured to go directly to my one extension.  My
> extension can register just fine.  My extension can dial out.  When I call
> my cell, the call connects and I have 2 way audio.  When I dial my DID from
> my cell, I can see the call hitting the FS server, but instead of ringing
> my extension, it goes straight to my extensions voicemail (which I can just
> fine).
>
>
>
> When I look the at the console, it appears (sorry if this is wrong, I'm
> only a day into free switch) that FS is attempting to route the call to my
> extension...@ my external ip?
>
>
>
> I see this line in the console:  2016-02-24 18:56:10.453983 [NOTICE]
> switch_channel.c:1101 New Channel sofia/internal/<extension>@
> *<externalip>*:46072
>
>
>
>
>
> Shouldn't that read sofia/internal/<extentsion>@<ip_of_extension>  ?
>
>
>
> Sofia Status says:
>
>
>
>                      Name        Type
>  Data  State
>
>
> =================================================================================================
>
>             external-ipv6   profile
> sip:mod_sofia@[::1]:5080            RUNNING (0)
>
>                  external      profile
> sip:mod_sofia@<external_ip>:5080     RUNNING (0)
>
> external::<uuid>          gateway                      sip:<flowrouteid>@
> sip.flowroute.com          REGED
>
>             internal-ipv6    profile
> sip:mod_sofia@[::1]:5060            RUNNING (0)
>
>                  internal       profile
> sip:mod_sofia@<internal_ip>:5060    RUNNING (0)
>
>
> =================================================================================================
>
>
>
> If I'm completely off base here, can anyone recommend where I can start
> looking to change/troubleshoot the issue?  I feel like it's just me missing
> something, I just can't determine what that might be.
>
>
>
> Thanks,
>
>   -Brian
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
>
> 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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Sat Feb 27 07:17:17 2016
From: aqsyounas at gmail.com (Aqs Younas)
Date: Sat, 27 Feb 2016 09:17:17 +0500
Subject: [Freeswitch-users] discriminate between session answered by
 users and voice mail automatic messages
In-Reply-To: <CABMX_w9f0uEAda4W5sEt=4W_NZBgKURBFN6K6XisYxteakhu7Q@mail.gmail.com>
References: <CAE6rfk1KG7S8eSgYM_Ydnpx9ix7d4bRQbThQ3OSSaCq_pDa3=w@mail.gmail.com>
	<CABMX_w9f0uEAda4W5sEt=4W_NZBgKURBFN6K6XisYxteakhu7Q@mail.gmail.com>
Message-ID: <CAGVOjVXFUdsHQRZf-8hdMg42_=pLS+WSeXtUP0a+eLMkRXj7PQ@mail.gmail.com>

Or listen for answer events and apply avmd on outbound leg.
On 26-Feb-2016 7:03 am, "?talo Rossi" <italo at freeswitch.org> wrote:

> https://freeswitch.org/confluence/display/FREESWITCH/mod_com_amd
>
> On Thu, Feb 25, 2016 at 7:24 PM, sharigo roma <sharigosrl at gmail.com>
> wrote:
>
>> Dear freeswitchers,
>>
>> is there a way to understand if a new session originated from a python
>> handler has been actually answered by the remote user or by the operator
>> voice mail with a message like "the customer you are trying to reach is not
>> available at the moment" ?
>>
>> The python code looks something like the following:
>>
>>
>> def handler(session, args):
>>  ## I do various things.... and than
>>
>>  session_string = str("sofia/gateway/MY_GW/0039123456789?)
>>  new_session = freeswitch.Session(session_string)
>>  if new_session.ready():
>>     # I get here also in case of automatic voice mail message
>>  else:
>>     # I supposed i?d get here in case of automatic ?unreachable? messages?
>>
>>
>> Thanks in advance
>> Lorenzo
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 ynasida at gmail.com  Sat Feb 27 11:36:46 2016
From: ynasida at gmail.com (=?UTF-8?B?0K7RgNC40Lkg0J3QsNGB0LjQtNCw?=)
Date: Sat, 27 Feb 2016 11:36:46 +0300
Subject: [Freeswitch-users] where mod_com_g729 ?
Message-ID: <CADihe3J3kv2RJ8DeueusW6aAkzGWNxR2h5zaKiLKV9asJg6LVg@mail.gmail.com>

Hi list

I have installed FS 1.6.6 via apt-get method (all modules) and don't see
mod_com_g729 anywhere.

Should I install it from source ?

Please advice.

# apt-cache search freeswitch | grep 729
freeswitch-mod-g729 - mod_g729 for FreeSWITCH
freeswitch-mod-g729-dbg - mod_g729 for FreeSWITCH (debug)
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From shabbirabbasi92 at gmail.com  Sat Feb 27 11:43:15 2016
From: shabbirabbasi92 at gmail.com (Shabbir abbasi)
Date: Sat, 27 Feb 2016 13:43:15 +0500
Subject: [Freeswitch-users] Exception: ReferenceError: File is not defined
	(near: "var fd = new File(logfile); 
Message-ID: <CAGpk5BQm1-aD7W4hAH=+AMr9VqeKL6ChwHdwWZgWsJEmB9siHQ@mail.gmail.com>

i am getting this error  in javascript  here is my code
logToFile.js

var logfile = "/usr/local/freeswitch/scripts/test.log";
var fd = new File(logfile);

i have defined but error is coming
 logToFile.js:3 Exception: ReferenceError: File is not defined (near: "var
fd = new File(logfile);")

Dont know why  ?
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From gregor at infomedia.si  Sat Feb 27 18:34:01 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Sat, 27 Feb 2016 16:34:01 +0100
Subject: [Freeswitch-users] originate and valet_park
Message-ID: <CAC7ttFHcL-X-yNxtQG-EkMB9vsiQmkd+iR-LAt+WiB7Wgo759g@mail.gmail.com>

Is it possible to originate call and after answer park it in valet? I tried
something like:
originate user/1000 &valet_park mylot 5555

Extension rings and after pickup call is disconnected.

Please for hint how to correctly format command or how to achive such
scenario in different way.

Best regards, Gregor
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From italo at freeswitch.org  Sat Feb 27 18:49:46 2016
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Sat, 27 Feb 2016 12:49:46 -0300
Subject: [Freeswitch-users] where mod_com_g729 ?
In-Reply-To: <CADihe3J3kv2RJ8DeueusW6aAkzGWNxR2h5zaKiLKV9asJg6LVg@mail.gmail.com>
References: <CADihe3J3kv2RJ8DeueusW6aAkzGWNxR2h5zaKiLKV9asJg6LVg@mail.gmail.com>
Message-ID: <CABMX_w-ga8GM9s+JHyWjSOAMLRqfJ=o5+YTLYww5V8qNbD7A4Q@mail.gmail.com>

https://freeswitch.org/confluence/display/FREESWITCH/mod_com_g729

On Sat, Feb 27, 2016 at 5:36 AM, ???? ?????? <ynasida at gmail.com> wrote:

> Hi list
>
> I have installed FS 1.6.6 via apt-get method (all modules) and don't see
> mod_com_g729 anywhere.
>
> Should I install it from source ?
>
> Please advice.
>
> # apt-cache search freeswitch | grep 729
> freeswitch-mod-g729 - mod_g729 for FreeSWITCH
> freeswitch-mod-g729-dbg - mod_g729 for FreeSWITCH (debug)
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From piotrek.gregor at gmail.com  Sat Feb 27 15:35:33 2016
From: piotrek.gregor at gmail.com (Peter Piotr)
Date: Sat, 27 Feb 2016 12:35:33 +0000
Subject: [Freeswitch-users] discriminate between session answered by
 users and voice mail automatic messages
In-Reply-To: <CAGVOjVXFUdsHQRZf-8hdMg42_=pLS+WSeXtUP0a+eLMkRXj7PQ@mail.gmail.com>
References: <CAE6rfk1KG7S8eSgYM_Ydnpx9ix7d4bRQbThQ3OSSaCq_pDa3=w@mail.gmail.com>
	<CABMX_w9f0uEAda4W5sEt=4W_NZBgKURBFN6K6XisYxteakhu7Q@mail.gmail.com>
	<CAGVOjVXFUdsHQRZf-8hdMg42_=pLS+WSeXtUP0a+eLMkRXj7PQ@mail.gmail.com>
Message-ID: <CAO3obxa8x9Y9=4ToA_jcXr6Szv+HF_VAAdFsZKqhy7H7EA+Unw@mail.gmail.com>

Module avmd will not help much in this case as this can be used to detect
sinusoidal tone of single frequency only (e.g. beep).

Piotr

On 27 February 2016 at 04:17, Aqs Younas <aqsyounas at gmail.com> wrote:

> Or listen for answer events and apply avmd on outbound leg.
>
>
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From avi at avimarcus.net  Sat Feb 27 20:28:00 2016
From: avi at avimarcus.net (Avi Marcus)
Date: Sat, 27 Feb 2016 17:28:00 +0000
Subject: [Freeswitch-users] Exception: ReferenceError: File is not
 defined (near: "var fd = new File(logfile); 
In-Reply-To: <CAGpk5BQm1-aD7W4hAH=+AMr9VqeKL6ChwHdwWZgWsJEmB9siHQ@mail.gmail.com>
References: <CAGpk5BQm1-aD7W4hAH=+AMr9VqeKL6ChwHdwWZgWsJEmB9siHQ@mail.gmail.com>
Message-ID: <0100015323c5d1c5-4b44a886-5859-4e6b-8caa-f29583a6a8f4-000000@email.amazonses.com>

The wiki seems to say that File isn't yet implemented for mod_v8...

-Avi Marcus
1-718-989-9485 (USA)
02-372-1570 (Israel)
020-3298-2875 (UK)

On Sat, Feb 27, 2016 at 10:43 AM, Shabbir abbasi <shabbirabbasi92 at gmail.com>
wrote:

> i am getting this error  in javascript  here is my code
> logToFile.js
>
> var logfile = "/usr/local/freeswitch/scripts/test.log";
> var fd = new File(logfile);
>
> i have defined but error is coming
>  logToFile.js:3 Exception: ReferenceError: File is not defined (near: "var
> fd = new File(logfile);")
>
> Dont know why  ?
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ynasida at gmail.com  Sat Feb 27 21:57:30 2016
From: ynasida at gmail.com (=?UTF-8?B?0K7RgNC40Lkg0J3QsNGB0LjQtNCw?=)
Date: Sat, 27 Feb 2016 21:57:30 +0300
Subject: [Freeswitch-users] where mod_com_g729 ?
In-Reply-To: <CABMX_w-ga8GM9s+JHyWjSOAMLRqfJ=o5+YTLYww5V8qNbD7A4Q@mail.gmail.com>
References: <CADihe3J3kv2RJ8DeueusW6aAkzGWNxR2h5zaKiLKV9asJg6LVg@mail.gmail.com>
	<CABMX_w-ga8GM9s+JHyWjSOAMLRqfJ=o5+YTLYww5V8qNbD7A4Q@mail.gmail.com>
Message-ID: <CADihe3JFLUtzLPQk6J+0rsVtVP5FG88uhbjyNeBOfhqP1brWfg@mail.gmail.com>

Looks like you didn't read my question.

2016-02-27 18:49 GMT+03:00 ?talo Rossi <italo at freeswitch.org>:

> https://freeswitch.org/confluence/display/FREESWITCH/mod_com_g729
>
> On Sat, Feb 27, 2016 at 5:36 AM, ???? ?????? <ynasida at gmail.com> wrote:
>
>> Hi list
>>
>> I have installed FS 1.6.6 via apt-get method (all modules) and don't see
>> mod_com_g729 anywhere.
>>
>> Should I install it from source ?
>>
>> Please advice.
>>
>> # apt-cache search freeswitch | grep 729
>> freeswitch-mod-g729 - mod_g729 for FreeSWITCH
>> freeswitch-mod-g729-dbg - mod_g729 for FreeSWITCH (debug)
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 brian at freeswitch.org  Sat Feb 27 22:11:04 2016
From: brian at freeswitch.org (Brian West)
Date: Sat, 27 Feb 2016 13:11:04 -0600
Subject: [Freeswitch-users] where mod_com_g729 ?
In-Reply-To: <CADihe3JFLUtzLPQk6J+0rsVtVP5FG88uhbjyNeBOfhqP1brWfg@mail.gmail.com>
References: <CADihe3J3kv2RJ8DeueusW6aAkzGWNxR2h5zaKiLKV9asJg6LVg@mail.gmail.com>
	<CABMX_w-ga8GM9s+JHyWjSOAMLRqfJ=o5+YTLYww5V8qNbD7A4Q@mail.gmail.com>
	<CADihe3JFLUtzLPQk6J+0rsVtVP5FG88uhbjyNeBOfhqP1brWfg@mail.gmail.com>
Message-ID: <CAEJMVkA=Zf-4tLo6p74kfNeSobBGkzR6ambz7i_hJYRep0mg+A@mail.gmail.com>

No he did read your question, there is no source to compile, did you read
the page ?talo sent?

On Saturday, February 27, 2016, ???? ?????? <ynasida at gmail.com> wrote:

> Looks like you didn't read my question.
>
> 2016-02-27 18:49 GMT+03:00 ?talo Rossi <italo at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','italo at freeswitch.org');>>:
>
>> https://freeswitch.org/confluence/display/FREESWITCH/mod_com_g729
>>
>> On Sat, Feb 27, 2016 at 5:36 AM, ???? ?????? <ynasida at gmail.com
>> <javascript:_e(%7B%7D,'cvml','ynasida at gmail.com');>> wrote:
>>
>>> Hi list
>>>
>>> I have installed FS 1.6.6 via apt-get method (all modules) and don't see
>>> mod_com_g729 anywhere.
>>>
>>> Should I install it from source ?
>>>
>>> Please advice.
>>>
>>> # apt-cache search freeswitch | grep 729
>>> freeswitch-mod-g729 - mod_g729 for FreeSWITCH
>>> freeswitch-mod-g729-dbg - mod_g729 for FreeSWITCH (debug)
>>>
>>>
>>> _________________________________________________________________________
>>> 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
>>>
>>
>>
>>
>> --
>> ?talo Rossi
>> italo at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','italo at 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


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

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 krice at freeswitch.org  Sun Feb 28 00:00:10 2016
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 27 Feb 2016 15:00:10 -0600
Subject: [Freeswitch-users] where mod_com_g729 ?
In-Reply-To: <CADihe3JFLUtzLPQk6J+0rsVtVP5FG88uhbjyNeBOfhqP1brWfg@mail.gmail.com>
References: <CADihe3J3kv2RJ8DeueusW6aAkzGWNxR2h5zaKiLKV9asJg6LVg@mail.gmail.com>
	<CABMX_w-ga8GM9s+JHyWjSOAMLRqfJ=o5+YTLYww5V8qNbD7A4Q@mail.gmail.com>
	<CADihe3JFLUtzLPQk6J+0rsVtVP5FG88uhbjyNeBOfhqP1brWfg@mail.gmail.com>
Message-ID: <DE690397-43CB-420F-BAC5-5E7DA57B81E3@freeswitch.org>

I'm pretty sure the link he gave you is the g729 commercial mod installation instructions and answers your question

Sent from my iPhone

> On Feb 27, 2016, at 12:57 PM, ???? ?????? <ynasida at gmail.com> wrote:
> 
> Looks like you didn't read my question.
> 
> 2016-02-27 18:49 GMT+03:00 ?talo Rossi <italo at freeswitch.org>:
>> https://freeswitch.org/confluence/display/FREESWITCH/mod_com_g729
>> 
>>> On Sat, Feb 27, 2016 at 5:36 AM, ???? ?????? <ynasida at gmail.com> wrote:
>>> Hi list
>>> 
>>> I have installed FS 1.6.6 via apt-get method (all modules) and don't see mod_com_g729 anywhere.
>>> 
>>> Should I install it from source ? 
>>> 
>>> Please advice.
>>> 
>>> # apt-cache search freeswitch | grep 729
>>> freeswitch-mod-g729 - mod_g729 for FreeSWITCH
>>> freeswitch-mod-g729-dbg - mod_g729 for FreeSWITCH (debug)
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 covici at ccs.covici.com  Sun Feb 28 00:54:40 2016
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sat, 27 Feb 2016 16:54:40 -0500
Subject: [Freeswitch-users] all-reg-options-ping and tls issue
In-Reply-To: <2E67ADAE1D3582409C90EBAE8C64C507122D361C@QSCDEMXP01a.ONE4ALL.LAN>
References: <2E67ADAE1D3582409C90EBAE8C64C5070149F6B3@QSCDEMXP01a.ONE4ALL.LAN>
	<6D3DEA47-DECB-4509-A327-B8DFEC1C1A83@kavun.ch>
	<2E67ADAE1D3582409C90EBAE8C64C507122D35C1@QSCDEMXP01a.ONE4ALL.LAN>
	<CAHtxdDdy-nq_gZDLKW5bE62-5TQV2u-fK=sSmVD5Y07qv=0Cxw@mail.gmail.com>
	<2E67ADAE1D3582409C90EBAE8C64C507122D35E9@QSCDEMXP01a.ONE4ALL.LAN>
	<CAHtxdDeAcZis=3SZnVwoK9j=29SLPTobDQUFGKSjLPfcFXGFTg@mail.gmail.com>
	<2E67ADAE1D3582409C90EBAE8C64C507122D361C@QSCDEMXP01a.ONE4ALL.LAN>
Message-ID: <22770.1456610080@ccs.covici.com>

Why use options ping at all -- I have never had good success with it, if
it fails then it causes lots of problems, more than its supposed
advantages.

Seeliger, Adam <Adam.Seeliger at qsc.de> wrote:

> Hi, here is the requested output.
> 
> I changed the real user, domain and ip address values into descriptions.
> I guess the values are not necessary?
> 
> Both, FreeSWITCH and the phone are in the same network (no NAT involved here)
> 
> Registrations:
> =================================================================================================
> Call-ID:         3134353634383833373032323631-ncwgvit2obfp
> User:            user at domain
> Contact:       "User Name" <sip:user at ip:60206;transport=tls;line=gvg9q8jh;fs_nat=yes;fs_path=sip%3Auser%40ip%3A60206>
> Agent:          snom715/8.7.5.35
> Status:         Registered(AUTO-NAT-2.0)(unknown) EXP(2016-02-26 16:21:32) EXPSECS(1912)
> Ping-Status:   Reachable
> Host:            hostname
> IP:               ip
> Port:            60206
> Auth-User:    user
> Auth-Realm: domain
> MWI-Account:          user at domain
> 
> Total items returned: 1
> =================================================================================================
> 2016-02-26 15:50:09.326648 [WARNING] sofia.c:5769 Sip user 'user at domain' is now Unreachable
> 2016-02-26 15:50:09.326648 [WARNING] sofia.c:5780 Expire sip user 'user at domain' due to options failure
> 
> As you can see, the user immediately got unregistered again ?
> 
> Best regards,
> Adam
> 
> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Sergey Safarov
> Gesendet: Freitag, 26. Februar 2016 15:45
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] all-reg-options-ping and tls issue
> 
> Please send output of command "sofia status profile internal reg <exten_number>"
> 
> On Fri, Feb 26, 2016 at 4:21 PM, Seeliger, Adam <Adam.Seeliger at qsc.de<mailto:Adam.Seeliger at qsc.de>> wrote:
> Hi,
> 
> the phone uses sip+tls.
> I test using a snom715, got plenty other phones here, but I guess they will behave the same way.
> It really looks like FreeSWITCH is doing something wrong (or is wrongly configured ? if there are any parameters for options ping supporting both, udp and tls)
> 
> Regards
> 
> Von: 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>] Im Auftrag von Sergey Safarov
> Gesendet: Freitag, 26. Februar 2016 13:59
> 
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] all-reg-options-ping and tls issue
> 
> If your phone has enabled SIPS uri please disable and use sip+tls.
> 
> 
> On Fri, Feb 26, 2016 at 3:37 PM, Seeliger, Adam <Adam.Seeliger at qsc.de<mailto:Adam.Seeliger at qsc.de>> wrote:
> Hi and thanks for the feedback,
> 
> sry that I did not respond for a long time.
> 
> I already use:
> <variable name=?sip-force-contact? value="NDLB-connectile-dysfunction-2.0"/>
> 
> I also tested all mentioned params below, nothing works.
> 
> When I register a User via TLS FreeSWITCH does not even try to ping the user.
> I turned sofia global siptrace on and watched the flow:
> 
> User                                        Server
> 13:09:33.311446:             REGISTER [TLS] ->
> 13:09:33.312552:             <- 401 UNAUTHORIZED [TLS]
> 13:09:33.331948:             REGISTER (AUTH) [TLS]  ->
> 13:09:33.336959:             <- 200 OK [TLS]
> Nothing happens
> 2016-02-26 13:10:00.619525 [WARNING] sofia.c:5769 Sip user 'user at host' is now Unreachable
> 2016-02-26 13:10:00.619525 [WARNING] sofia.c:5780 Expire sip user 'user at host' due to options failure
> 
> When I REGISTER the User via UDP FreeSWITCH starts to ping (OPTIONS) the user as soon as he is registered.
> 
> Is there any way to force FreeSWITCH to send OPTIONs in both, udp and tls, depending on the registration?
> 
> Thanks in advance,
> Adam
> 
> Von: 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>] Im Auftrag von Emrah
> Gesendet: Freitag, 29. Januar 2016 09:25
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] all-reg-options-ping and tls issue
> 
> Hi!
> This is interesting. I experienced something rather similar where calls would drop because FS would timeout on certain packets sent over UDP instead of TLS.
> I assume you mean FS exits with port 5060 instead of port 5061? Because the port on the remote end should be dynamically set.
> I found out that in my case, what works best even with TLS, is to use:
> <variable name=?sip-force-contact? value="NDLB-connectile-dysfunction-2.0"/>
> This goes as far as it can to lay out the path to contacting the client with all consideration in regards to NAT and dynamic ports.
> Not sure if it will help you. I?ve personally disabled options-ping an let my clients deal with keep-alive instead.
> 
> You could also look into:
>     <!-- add a ;received="<ip>:<port>" to the contact when replying to register for nat handling -->
> <!? <param name="NDLB-received-in-nat-reg-contact" value="true"/> -->
> <!? <param name="bind-params" value="transport=tcp,udp,tls"/>  -->
> <!-- additional bind parameters for TLS -->
>     <param name=?tls-bind-params" value="transport=tls"/>
> 
> I?ll leave it up to you to investigate those options more in details on the FS documentation.
> 
> Please keep us posted!
> 
> E
> On Jan 28, 2016, at 11:48 AM, Seeliger, Adam <Adam.Seeliger at qsc.de<mailto:Adam.Seeliger at qsc.de>> wrote:
> 
> Hi all,
> 
> I have a problem, when I enable TLS and register a phone using TLS on Port 5061.
> FreeSWITCH still tries to ?ping? the phone using Port 5060 using UDP, which is ignored by the phone.
> Moments later FreeSWITCH deletes the registration, because ?unregister-on-options-fail? is set to ?true?.
> 
> I already figured out, that you can set ?all-reg-options-ping? to ?udp-only?, but this would completely disable this feature for TLS.
> Is there any way to ping TLS registered using TLS?
> 
> Thanks in advance
> 
> - Adam
> 
> _________________________________________________________________________
> 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
> 
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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

         John Covici
         covici at ccs.covici.com


From max at nysolutions.com  Sun Feb 28 03:58:55 2016
From: max at nysolutions.com (Moishe Grunstein)
Date: Sun, 28 Feb 2016 00:58:55 +0000
Subject: [Freeswitch-users] Inbound calls mis-routing routing to
 internal extension with external IP?
In-Reply-To: <CAA_Md=v943_=QbEWo4BxPK7g4UNnPkFN97GaEn6mwZie6aeoAA@mail.gmail.com>
References: <CAA_Md=uAGmnqLkY3g128QcjoBis4VnoTwSUEZpqAws35AwRsHQ@mail.gmail.com>
	<23b1626b3814448b90842a279dc30be5@nysolutions.com>
	<CAA_Md=t4NR-ocAsnvOmD39XGNAexYVZWsWy4taYbiXU7-N1qEg@mail.gmail.com>
	<CAEJMVkC+=U50t2nEYrn9CU1OQnXOUcVXKmpzaB8KF=Gbh7T-BQ@mail.gmail.com>
	<CAA_Md=v=PGrR_0s90SK_ijZO5eScqJYN+Utqu22nwqbZEQXvLA@mail.gmail.com>
	<CAA_Md=sTVGHHmAvtV9EV2OvBuLkdnQ3b9wRaSCMYjBwf71p-yA@mail.gmail.com>
	<bf0ede9ef70640818f0607ceb46f5434@nysolutions.com>
	<CAA_Md=tJGhw7_BGz4qFhXAz_dnpy8d7+2L0cBRm6ZaF9jPeHaQ@mail.gmail.com>
	<62e0a3e6c4e14d40b15aedb1d3e1b3b6@nysolutions.com>
	<CAA_Md=v943_=QbEWo4BxPK7g4UNnPkFN97GaEn6mwZie6aeoAA@mail.gmail.com>
Message-ID: <ee347416888442bb99d75ffde4f66930@nysolutions.com>

You should not need to configure an acl for your cidr unless your endpoints are not registering.
The only acl you would need is for your sip providers ip?s if they are ip authenticating and sending traffic to your internal profile.

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 Brian Chow
Sent: Friday, February 26, 2016 5:45 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP?

I think that solved it, thanks!  I did have ACL configured, just apparently incorrectly.  Out of the (fusionpbx) box I had two ACL lists, one called LAN with a default of allow, which I had entered my CIDR into.  One called domain, which was default deny, but I had made an allow entry for my domain, but no CIDR.  When I added my CIDR to the domain list, it started working.

Thanks again for walking me through the troubleshooting, sorry for such a noob mistake :)

On Thu, Feb 25, 2016 at 5:50 PM, Moishe Grunstein <max at nysolutions.com<mailto:max at nysolutions.com>> wrote:
Can you show a log of a internal call?
Are the phones showing registered? Show registrations
Did you by any chance add your ip range to the ACL.


Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650<tel:212.400.7650> 888.IPPBX.US<http://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> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Brian Chow
Sent: Thursday, February 25, 2016 8:18 PM

To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP?

Moishe,
  Thanks for double checking this, I hadn't thought to try an extension <-> extension call as it's a private server just for me, however, after creating another extension, I cannot make internal calls.  It seems, I can only make outbound calls.

Internal -> External is working.
External -> Internal only works to the FS server, not to the extension.
Internal -> Internal does not work.



On Thu, Feb 25, 2016 at 8:12 AM, Moishe Grunstein <max at nysolutions.com<mailto:max at nysolutions.com>> wrote:
Does internal call also go direct to voicemail?
Are you sure you don?t have a *99 before the extension number?

Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650<tel:212.400.7650> 888.IPPBX.US<http://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> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Brian Chow
Sent: Thursday, February 25, 2016 11:00 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP?


Ok, in that case, what do I do next to figure out why inbound calls go directly to voicemail instead of ringing the extension?
On Feb 25, 2016 7:57 AM, "Brian Chow" <brianchow at digitaldescent.net<mailto:brianchow at digitaldescent.net>> wrote:

Ok, in that case, what do I look at next to see why it goes straight to voicemail instead of ringing my extension?
On Feb 25, 2016 5:52 AM, "Brian West" <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:
Its just the channel name so don't get tripped up by that, its just named with some default data about the session.

On Wed, Feb 24, 2016 at 11:33 PM, Brian Chow <freeswitch at digitaldescent.net<mailto:freeswitch at digitaldescent.net>> wrote:
Debug console output: https://pastebin.freeswitch.org/24572

My fqdn only resolves inside the network, but the external-ip is configured in FS to use stun, which reports my external ip.

>Is your endpoint registered? How often is it set to reregister?
Yes, and the zoiper default registration expiry is 3600

>Does your router have a sip alg? Are the ports opened in your firewall?
> ish, no GUI for it, but I did make sure to disable it from the command line (Ubiquiti Edgerouter).  No firewall ports are explicitly open, but allow bi-directional with existing states.  I do get 2 way audio when calling out, and I also can hear the system forwarding my call in to voicemail; the voicemail message and prompts.  I know asterisk is different from FS, but my asterisk experience has proven that at least with asterisk, port forwarding isn't needed.  Also, unfortunately, I cannot get a static ip.

Thanks!
  -Brian

On Wed, Feb 24, 2016 at 8:31 PM, Moishe Grunstein <max at nysolutions.com<mailto:max at nysolutions.com>> wrote:
It is probably sending call to your extension at domainname, if your external ip is your domain name then you will see the external ip.
Is your endpoint registered? How often is it set to reregister?
Does your router have a sip alg? Are the ports opened in your firewall?
Very hard to guess without a log of a call with debug enabled.
sofia global sip trace on

Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650<tel:212.400.7650> 888.IPPBX.US<http://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> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Brian Chow
Sent: Wednesday, February 24, 2016 10:00 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP?

Hello all!
  I'm new to freeswitch, so I'm sure this is just a newbie configuration error.  Sorry if it's been answered a million times, my google searches always just bring up the standard NAT configuration pages.  I've already followed the confluence page on configuring NAT.

My Setup:
Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3 instance.
SIP Provider: Flowroute

NAT: extension and freeswitch are on the same network, both of which are behind NAT and connecting to flowroute.  I configured external sip and rtp to use stun entries.  External profile is using $${external_rtp/sip_ip} for ext-rtp/sip respectively.

I have one DID configured to go directly to my one extension.  My extension can register just fine.  My extension can dial out.  When I call my cell, the call connects and I have 2 way audio.  When I dial my DID from my cell, I can see the call hitting the FS server, but instead of ringing my extension, it goes straight to my extensions voicemail (which I can just fine).

When I look the at the console, it appears (sorry if this is wrong, I'm only a day into free switch) that FS is attempting to route the call to my extension...@ my external ip?

I see this line in the console:  2016-02-24 18:56:10.453983 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/<extension>@<externalip>:46072


Shouldn't that read sofia/internal/<extentsion>@<ip_of_extension>  ?

Sofia Status says:

                     Name        Type                                      Data  State
=================================================================================================
            external-ipv6   profile                   sip:mod_sofia@[::1]:5080            RUNNING (0)
                 external      profile           sip:mod_sofia@<external_ip>:5080<sip:mod_sofia@%3cexternal_ip%3e:5080>     RUNNING (0)
external::<uuid>          gateway                      sip:<flowrouteid>@sip.flowroute.com<http://sip.flowroute.com>          REGED
            internal-ipv6    profile                   sip:mod_sofia@[::1]:5060            RUNNING (0)
                 internal       profile            sip:mod_sofia@<internal_ip>:5060<sip:mod_sofia@%3cinternal_ip%3e:5060>    RUNNING (0)
=================================================================================================

If I'm completely off base here, can anyone recommend where I can start looking to change/troubleshoot the issue?  I feel like it's just me missing something, I just can't determine what that might be.

Thanks,
  -Brian

_________________________________________________________________________
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
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http://www.freeswitchsolutions.com

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

Brian West
brian at freeswitch.org<mailto:brian at freeswitch.org>

[http://billing.freeswitch.org/templates/default/img/whmcslogo.png]

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

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

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iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest

_________________________________________________________________________
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
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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
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_________________________________________________________________________
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
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From freeswitch at digitaldescent.net  Sun Feb 28 06:03:41 2016
From: freeswitch at digitaldescent.net (Brian Chow)
Date: Sat, 27 Feb 2016 19:03:41 -0800
Subject: [Freeswitch-users] Inbound calls mis-routing routing to
 internal extension with external IP?
In-Reply-To: <ee347416888442bb99d75ffde4f66930@nysolutions.com>
References: <CAA_Md=uAGmnqLkY3g128QcjoBis4VnoTwSUEZpqAws35AwRsHQ@mail.gmail.com>
	<23b1626b3814448b90842a279dc30be5@nysolutions.com>
	<CAA_Md=t4NR-ocAsnvOmD39XGNAexYVZWsWy4taYbiXU7-N1qEg@mail.gmail.com>
	<CAEJMVkC+=U50t2nEYrn9CU1OQnXOUcVXKmpzaB8KF=Gbh7T-BQ@mail.gmail.com>
	<CAA_Md=v=PGrR_0s90SK_ijZO5eScqJYN+Utqu22nwqbZEQXvLA@mail.gmail.com>
	<CAA_Md=sTVGHHmAvtV9EV2OvBuLkdnQ3b9wRaSCMYjBwf71p-yA@mail.gmail.com>
	<bf0ede9ef70640818f0607ceb46f5434@nysolutions.com>
	<CAA_Md=tJGhw7_BGz4qFhXAz_dnpy8d7+2L0cBRm6ZaF9jPeHaQ@mail.gmail.com>
	<62e0a3e6c4e14d40b15aedb1d3e1b3b6@nysolutions.com>
	<CAA_Md=v943_=QbEWo4BxPK7g4UNnPkFN97GaEn6mwZie6aeoAA@mail.gmail.com>
	<ee347416888442bb99d75ffde4f66930@nysolutions.com>
Message-ID: <CAA_Md=tomE9AAfCt0cu-iy4x72EeGMQSUkCRPm1+1_9At8x7NQ@mail.gmail.com>

The extensions were always registering. The clients reported as registered,
and I was able to make outbound calls without issue.  It was only inbound
calls that went directly to voicemail instead of ringing. I also could not
call extension to extension. After adding the cidr to the acl list both
internal extension calls and inbound calls were working. However, it sounds
like you believe that should not be the case based on that change?

I can post config samples later when I get back home.  Or at least
fusionpbx screenshots.  The I'm not sure where fusionpbx gets all the
settings from. The internal and external profiles all have .noload
extensions, which seems different from the other FreeSWITCH config files.
On Feb 27, 2016 5:01 PM, "Moishe Grunstein" <max at nysolutions.com> wrote:

> You should not need to configure an acl for your cidr unless your
> endpoints are not registering.
>
> The only acl you would need is for your sip providers ip?s if they are ip
> authenticating and sending traffic to your internal profile.
>
>
>
> 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 *Brian Chow
> *Sent:* Friday, February 26, 2016 5:45 PM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Inbound calls mis-routing routing to
> internal extension with external IP?
>
>
>
> I think that solved it, thanks!  I did have ACL configured, just
> apparently incorrectly.  Out of the (fusionpbx) box I had two ACL lists,
> one called LAN with a default of allow, which I had entered my CIDR into.
> One called domain, which was default deny, but I had made an allow entry
> for my domain, but no CIDR.  When I added my CIDR to the domain list, it
> started working.
>
>
>
> Thanks again for walking me through the troubleshooting, sorry for such a
> noob mistake :)
>
>
>
> On Thu, Feb 25, 2016 at 5:50 PM, Moishe Grunstein <max at nysolutions.com>
> wrote:
>
> Can you show a log of a internal call?
>
> Are the phones showing registered? Show registrations
>
> Did you by any chance add your ip range to the ACL.
>
>
>
>
>
> 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 *Brian Chow
> *Sent:* Thursday, February 25, 2016 8:18 PM
>
>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Inbound calls mis-routing routing to
> internal extension with external IP?
>
>
>
> Moishe,
>
>   Thanks for double checking this, I hadn't thought to try an extension
> <-> extension call as it's a private server just for me, however, after
> creating another extension, I cannot make internal calls.  It seems, I can
> only make outbound calls.
>
>
>
> Internal -> External is working.
>
> External -> Internal only works to the FS server, not to the extension.
>
> Internal -> Internal does not work.
>
>
>
>
>
>
>
> On Thu, Feb 25, 2016 at 8:12 AM, Moishe Grunstein <max at nysolutions.com>
> wrote:
>
> Does internal call also go direct to voicemail?
>
> Are you sure you don?t have a *99 before the extension number?
>
>
>
> 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 *Brian Chow
> *Sent:* Thursday, February 25, 2016 11:00 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Inbound calls mis-routing routing to
> internal extension with external IP?
>
>
>
> Ok, in that case, what do I do next to figure out why inbound calls go
> directly to voicemail instead of ringing the extension?
>
> On Feb 25, 2016 7:57 AM, "Brian Chow" <brianchow at digitaldescent.net>
> wrote:
>
> Ok, in that case, what do I look at next to see why it goes straight to
> voicemail instead of ringing my extension?
>
> On Feb 25, 2016 5:52 AM, "Brian West" <brian at freeswitch.org> wrote:
>
> Its just the channel name so don't get tripped up by that, its just named
> with some default data about the session.
>
>
>
> On Wed, Feb 24, 2016 at 11:33 PM, Brian Chow <
> freeswitch at digitaldescent.net> wrote:
>
> Debug console output: https://pastebin.freeswitch.org/24572
>
>
>
> My fqdn only resolves inside the network, but the external-ip is
> configured in FS to use stun, which reports my external ip.
>
>
>
> >Is your endpoint registered? How often is it set to reregister?
>
> Yes, and the zoiper default registration expiry is 3600
>
>
>
> >Does your router have a sip alg? Are the ports opened in your firewall?
>
> > ish, no GUI for it, but I did make sure to disable it from the command
> line (Ubiquiti Edgerouter).  No firewall ports are explicitly open, but
> allow bi-directional with existing states.  I do get 2 way audio when
> calling out, and I also can hear the system forwarding my call in to
> voicemail; the voicemail message and prompts.  I know asterisk is different
> from FS, but my asterisk experience has proven that at least with asterisk,
> port forwarding isn't needed.  Also, unfortunately, I cannot get a static
> ip.
>
>
>
> Thanks!
>
>   -Brian
>
>
>
> On Wed, Feb 24, 2016 at 8:31 PM, Moishe Grunstein <max at nysolutions.com>
> wrote:
>
> It is probably sending call to your extension at domainname, if your
> external ip is your domain name then you will see the external ip.
>
> Is your endpoint registered? How often is it set to reregister?
>
> Does your router have a sip alg? Are the ports opened in your firewall?
>
> Very hard to guess without a log of a call with debug enabled.
>
> sofia global sip trace on
>
>
>
> 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 *Brian Chow
> *Sent:* Wednesday, February 24, 2016 10:00 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Inbound calls mis-routing routing to
> internal extension with external IP?
>
>
>
> Hello all!
>
>   I'm new to freeswitch, so I'm sure this is just a newbie configuration
> error.  Sorry if it's been answered a million times, my google searches
> always just bring up the standard NAT configuration pages.  I've already
> followed the confluence page on configuring NAT.
>
>
>
> My Setup:
>
> Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3
> instance.
>
> SIP Provider: Flowroute
>
>
>
> NAT: extension and freeswitch are on the same network, both of which are
> behind NAT and connecting to flowroute.  I configured external sip and rtp
> to use stun entries.  External profile is using $${external_rtp/sip_ip} for
> ext-rtp/sip respectively.
>
>
>
> I have one DID configured to go directly to my one extension.  My
> extension can register just fine.  My extension can dial out.  When I call
> my cell, the call connects and I have 2 way audio.  When I dial my DID from
> my cell, I can see the call hitting the FS server, but instead of ringing
> my extension, it goes straight to my extensions voicemail (which I can just
> fine).
>
>
>
> When I look the at the console, it appears (sorry if this is wrong, I'm
> only a day into free switch) that FS is attempting to route the call to my
> extension...@ my external ip?
>
>
>
> I see this line in the console:  2016-02-24 18:56:10.453983 [NOTICE]
> switch_channel.c:1101 New Channel sofia/internal/<extension>@
> *<externalip>*:46072
>
>
>
>
>
> Shouldn't that read sofia/internal/<extentsion>@<ip_of_extension>  ?
>
>
>
> Sofia Status says:
>
>
>
>                      Name        Type
>  Data  State
>
>
> =================================================================================================
>
>             external-ipv6   profile
> sip:mod_sofia@[::1]:5080            RUNNING (0)
>
>                  external      profile
> sip:mod_sofia@<external_ip>:5080     RUNNING (0)
>
> external::<uuid>          gateway                      sip:<flowrouteid>@
> sip.flowroute.com          REGED
>
>             internal-ipv6    profile
> sip:mod_sofia@[::1]:5060            RUNNING (0)
>
>                  internal       profile
> sip:mod_sofia@<internal_ip>:5060    RUNNING (0)
>
>
> =================================================================================================
>
>
>
> If I'm completely off base here, can anyone recommend where I can start
> looking to change/troubleshoot the issue?  I feel like it's just me missing
> something, I just can't determine what that might be.
>
>
>
> Thanks,
>
>   -Brian
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 max at nysolutions.com  Sun Feb 28 06:47:48 2016
From: max at nysolutions.com (Moishe Grunstein)
Date: Sun, 28 Feb 2016 03:47:48 +0000
Subject: [Freeswitch-users] Inbound calls mis-routing routing to
 internal extension with external IP?
In-Reply-To: <CAA_Md=tomE9AAfCt0cu-iy4x72EeGMQSUkCRPm1+1_9At8x7NQ@mail.gmail.com>
References: <CAA_Md=uAGmnqLkY3g128QcjoBis4VnoTwSUEZpqAws35AwRsHQ@mail.gmail.com>
	<23b1626b3814448b90842a279dc30be5@nysolutions.com>
	<CAA_Md=t4NR-ocAsnvOmD39XGNAexYVZWsWy4taYbiXU7-N1qEg@mail.gmail.com>
	<CAEJMVkC+=U50t2nEYrn9CU1OQnXOUcVXKmpzaB8KF=Gbh7T-BQ@mail.gmail.com>
	<CAA_Md=v=PGrR_0s90SK_ijZO5eScqJYN+Utqu22nwqbZEQXvLA@mail.gmail.com>
	<CAA_Md=sTVGHHmAvtV9EV2OvBuLkdnQ3b9wRaSCMYjBwf71p-yA@mail.gmail.com>
	<bf0ede9ef70640818f0607ceb46f5434@nysolutions.com>
	<CAA_Md=tJGhw7_BGz4qFhXAz_dnpy8d7+2L0cBRm6ZaF9jPeHaQ@mail.gmail.com>
	<62e0a3e6c4e14d40b15aedb1d3e1b3b6@nysolutions.com>
	<CAA_Md=v943_=QbEWo4BxPK7g4UNnPkFN97GaEn6mwZie6aeoAA@mail.gmail.com>
	<ee347416888442bb99d75ffde4f66930@nysolutions.com>
	<CAA_Md=tomE9AAfCt0cu-iy4x72EeGMQSUkCRPm1+1_9At8x7NQ@mail.gmail.com>
Message-ID: <f9b62c614f3d498092aaf5420e7bb865@nysolutions.com>

Your problem is what you initially added to the ACL, if you remove it all and reload ACL, it should work fine.

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 Brian Chow
Sent: Saturday, February 27, 2016 10:04 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP?


The extensions were always registering. The clients reported as registered, and I was able to make outbound calls without issue.  It was only inbound calls that went directly to voicemail instead of ringing. I also could not call extension to extension. After adding the cidr to the acl list both internal extension calls and inbound calls were working. However, it sounds like you believe that should not be the case based on that change?

I can post config samples later when I get back home.  Or at least fusionpbx screenshots.  The I'm not sure where fusionpbx gets all the settings from. The internal and external profiles all have .noload extensions, which seems different from the other FreeSWITCH config files.
On Feb 27, 2016 5:01 PM, "Moishe Grunstein" <max at nysolutions.com<mailto:max at nysolutions.com>> wrote:
You should not need to configure an acl for your cidr unless your endpoints are not registering.
The only acl you would need is for your sip providers ip?s if they are ip authenticating and sending traffic to your internal profile.

Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650<tel:212.400.7650> 888.IPPBX.US<http://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> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Brian Chow
Sent: Friday, February 26, 2016 5:45 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP?

I think that solved it, thanks!  I did have ACL configured, just apparently incorrectly.  Out of the (fusionpbx) box I had two ACL lists, one called LAN with a default of allow, which I had entered my CIDR into.  One called domain, which was default deny, but I had made an allow entry for my domain, but no CIDR.  When I added my CIDR to the domain list, it started working.

Thanks again for walking me through the troubleshooting, sorry for such a noob mistake :)

On Thu, Feb 25, 2016 at 5:50 PM, Moishe Grunstein <max at nysolutions.com<mailto:max at nysolutions.com>> wrote:
Can you show a log of a internal call?
Are the phones showing registered? Show registrations
Did you by any chance add your ip range to the ACL.


Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650<tel:212.400.7650> 888.IPPBX.US<http://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> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Brian Chow
Sent: Thursday, February 25, 2016 8:18 PM

To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP?

Moishe,
  Thanks for double checking this, I hadn't thought to try an extension <-> extension call as it's a private server just for me, however, after creating another extension, I cannot make internal calls.  It seems, I can only make outbound calls.

Internal -> External is working.
External -> Internal only works to the FS server, not to the extension.
Internal -> Internal does not work.



On Thu, Feb 25, 2016 at 8:12 AM, Moishe Grunstein <max at nysolutions.com<mailto:max at nysolutions.com>> wrote:
Does internal call also go direct to voicemail?
Are you sure you don?t have a *99 before the extension number?

Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650<tel:212.400.7650> 888.IPPBX.US<http://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> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Brian Chow
Sent: Thursday, February 25, 2016 11:00 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP?


Ok, in that case, what do I do next to figure out why inbound calls go directly to voicemail instead of ringing the extension?
On Feb 25, 2016 7:57 AM, "Brian Chow" <brianchow at digitaldescent.net<mailto:brianchow at digitaldescent.net>> wrote:

Ok, in that case, what do I look at next to see why it goes straight to voicemail instead of ringing my extension?
On Feb 25, 2016 5:52 AM, "Brian West" <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:
Its just the channel name so don't get tripped up by that, its just named with some default data about the session.

On Wed, Feb 24, 2016 at 11:33 PM, Brian Chow <freeswitch at digitaldescent.net<mailto:freeswitch at digitaldescent.net>> wrote:
Debug console output: https://pastebin.freeswitch.org/24572

My fqdn only resolves inside the network, but the external-ip is configured in FS to use stun, which reports my external ip.

>Is your endpoint registered? How often is it set to reregister?
Yes, and the zoiper default registration expiry is 3600

>Does your router have a sip alg? Are the ports opened in your firewall?
> ish, no GUI for it, but I did make sure to disable it from the command line (Ubiquiti Edgerouter).  No firewall ports are explicitly open, but allow bi-directional with existing states.  I do get 2 way audio when calling out, and I also can hear the system forwarding my call in to voicemail; the voicemail message and prompts.  I know asterisk is different from FS, but my asterisk experience has proven that at least with asterisk, port forwarding isn't needed.  Also, unfortunately, I cannot get a static ip.

Thanks!
  -Brian

On Wed, Feb 24, 2016 at 8:31 PM, Moishe Grunstein <max at nysolutions.com<mailto:max at nysolutions.com>> wrote:
It is probably sending call to your extension at domainname, if your external ip is your domain name then you will see the external ip.
Is your endpoint registered? How often is it set to reregister?
Does your router have a sip alg? Are the ports opened in your firewall?
Very hard to guess without a log of a call with debug enabled.
sofia global sip trace on

Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650<tel:212.400.7650> 888.IPPBX.US<http://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> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Brian Chow
Sent: Wednesday, February 24, 2016 10:00 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP?

Hello all!
  I'm new to freeswitch, so I'm sure this is just a newbie configuration error.  Sorry if it's been answered a million times, my google searches always just bring up the standard NAT configuration pages.  I've already followed the confluence page on configuring NAT.

My Setup:
Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3 instance.
SIP Provider: Flowroute

NAT: extension and freeswitch are on the same network, both of which are behind NAT and connecting to flowroute.  I configured external sip and rtp to use stun entries.  External profile is using $${external_rtp/sip_ip} for ext-rtp/sip respectively.

I have one DID configured to go directly to my one extension.  My extension can register just fine.  My extension can dial out.  When I call my cell, the call connects and I have 2 way audio.  When I dial my DID from my cell, I can see the call hitting the FS server, but instead of ringing my extension, it goes straight to my extensions voicemail (which I can just fine).

When I look the at the console, it appears (sorry if this is wrong, I'm only a day into free switch) that FS is attempting to route the call to my extension...@ my external ip?

I see this line in the console:  2016-02-24 18:56:10.453983 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/<extension>@<externalip>:46072


Shouldn't that read sofia/internal/<extentsion>@<ip_of_extension>  ?

Sofia Status says:

                     Name        Type                                      Data  State
=================================================================================================
            external-ipv6   profile                   sip:mod_sofia@[::1]:5080            RUNNING (0)
                 external      profile           sip:mod_sofia@<external_ip>:5080<sip:mod_sofia@%3cexternal_ip%3e:5080>     RUNNING (0)
external::<uuid>          gateway                      sip:<flowrouteid>@sip.flowroute.com<http://sip.flowroute.com>          REGED
            internal-ipv6    profile                   sip:mod_sofia@[::1]:5060            RUNNING (0)
                 internal       profile            sip:mod_sofia@<internal_ip>:5060<sip:mod_sofia@%3cinternal_ip%3e:5060>    RUNNING (0)
=================================================================================================

If I'm completely off base here, can anyone recommend where I can start looking to change/troubleshoot the issue?  I feel like it's just me missing something, I just can't determine what that might be.

Thanks,
  -Brian

_________________________________________________________________________
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[http://billing.freeswitch.org/templates/default/img/whmcslogo.png]

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

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

Official FreeSWITCH Sites
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_________________________________________________________________________
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http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
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Official FreeSWITCH Sites
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From mgiammarco at gmail.com  Sun Feb 28 14:00:31 2016
From: mgiammarco at gmail.com (Mario Giammarco)
Date: Sun, 28 Feb 2016 11:00:31 +0000 (UTC)
Subject: [Freeswitch-users] Recording and mono directional webrtc video
References: <loom.20160222T182837-828@post.gmane.org>
Message-ID: <loom.20160228T115943-266@post.gmane.org>

Mario Giammarco <mgiammarco at ...> writes:

> I need a bidirectional audio 
> call with one direction video only from caller. And I need to record audio 
> and video from caller.

Hello,
nobody can help me? I am asking only if it is possible and supported.

Thanks,
Mario



From shlomis at liveperson.com  Sun Feb 28 18:17:55 2016
From: shlomis at liveperson.com (Shlomi Schwartz)
Date: Sun, 28 Feb 2016 17:17:55 +0200
Subject: [Freeswitch-users] MOD_VERTO Documentation
Message-ID: <CA+bkYdFUrCJyHn5FyYBmuvhiDDLPxv5GpBAxs5hzy4aernKOEA@mail.gmail.com>

Hi all,

Hello, I'm trying to implement mod_verto JSONRPC on IOS, is there any
documentation regarding the dialogParams on verto.invite method?

The only documentation I found was :
https://freeswitch.org/confluence/display/FREESWITCH/mod_verto

Thanks :)

-- 
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R&D Technical Leader
T: +972-74-700-4511
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<http://www.facebook.com/LivePersonInc> We Create Meaningful Connections
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This message may contain confidential and/or privileged information. 
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From italo at freeswitch.org  Sun Feb 28 20:14:17 2016
From: italo at freeswitch.org (=?utf-8?q?=C3=8Dtalo_Rossi?=)
Date: Sun, 28 Feb 2016 09:14:17 -0800 (PST)
Subject: [Freeswitch-users] MOD_VERTO Documentation
In-Reply-To: <CA+bkYdFUrCJyHn5FyYBmuvhiDDLPxv5GpBAxs5hzy4aernKOEA@mail.gmail.com>
References: <CA+bkYdFUrCJyHn5FyYBmuvhiDDLPxv5GpBAxs5hzy4aernKOEA@mail.gmail.com>
Message-ID: <a2mogxx2kvmcxtd607rh3qbxf-0@mailer.nylas.com>

Hi Shlomi,

  

We have a detailed doc here, but using the javascript lib, maybe it can give
you some light, take a look at http://evoluxbr.github.io/verto-docs/

> On Feb 28 2016, at 1:08 pm, Shlomi Schwartz &lt;shlomis at liveperson.com&gt;
wrote:  

>

> Hi all,

>

>  

>

> Hello, I'm trying to implement mod_verto JSONRPC on IOS, is there any
documentation regarding the dialogParams on verto.invite method?

>

>  

>

> The only documentation I found was :
<https://freeswitch.org/confluence/display/FREESWITCH/mod_verto>

>

>  

>

> Thanks :)  

>

>  

>

> \--  

>

> ![](https://signature.s3.amazonaws.com/2015/lp_logo.png)  
> ---  
> Shlomi Schwartz  
> R&amp;D Technical Leader  
> T: +972-74-700-4511  
> |  | [![](https://signature.s3.amazonaws.com/2015/LinkedIn.png)](http://www.
linkedin.com/company/164748) | [![](https://signature.s3.amazonaws.com/2015/Tw
itter.png)](http://twitter.com/liveperson) | [![](https://signature.s3.amazona
ws.com/2015/Facebook.png)](http://www.facebook.com/LivePersonInc)  
> ---|---|---  
> We Create Meaningful Connections  
> [![](https://signature.s3.amazonaws.com/2015/banners/Inc_email_banner_2.8.20
16.jpg)](http://info.liveperson.com/201602/liz-welch/liveperson-robert-
locascio-on-death-of-the-800-call.html)  
>  
>  

>

> This message may contain confidential and/or privileged information.

>

> If you are not the addressee or authorized to receive this on behalf of the
addressee you must not use, copy, disclose or take action based on this
message or any information herein.

>

> If you have received this message in error, please advise the sender
immediately by reply email and delete this message. Thank you.

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From lists at kavun.ch  Sun Feb 28 21:33:49 2016
From: lists at kavun.ch (Emrah)
Date: Sun, 28 Feb 2016 19:33:49 +0100
Subject: [Freeswitch-users] SRTP breaks my TLS session
In-Reply-To: <93C84919-0D70-4DD3-BB86-46DE87492F9C@kavun.ch>
References: <84F64031-B943-4671-A0BC-66FC02DF7C51@kavun.ch>
	<CAEJMVkCsxNUfWNBAkJQp5k86Nb+1=vFBj13XKEc+ad1_Hst8og@mail.gmail.com>
	<88ACC7C7-589C-4CC8-A146-3DBF31B84A2E@mgtech.com>
	<93C84919-0D70-4DD3-BB86-46DE87492F9C@kavun.ch>
Message-ID: <14EF2D3F-9034-4D59-BC37-16B6E20BAEE8@kavun.ch>

Hello there,
I can confirm that a PCAP gives me a bunch of TLS packets.
How do you suggest debugging this?

Thanks!
> On Feb 26, 2016, at 10:36 AM, Emrah <lists at kavun.ch> wrote:
> 
> Thanks for this. 
> 
> This isn?t just a yealink thing. I?ve encountered sporadic issues with soft phones and other desk phones as well.
> 
> I didn?t use the PCap capture feature because I had assumed it would give me a bunch of TLS packets. I?ll test that and revert back.
> 
> How can we explain that I have more calls failing if I register multiple accounts?
> 
> Emrah
> 
>> On Feb 26, 2016, at 6:37 AM, Mario G <mario_fs at mgtech.com <mailto:mario_fs at mgtech.com>> wrote:
>> 
>> You may want to run a pcap trace on the Yealink. It?s under settings->Configuration. Start/test/export.
>> 
>>> On Feb 25, 2016, at 3:35 PM, Brian West <brian at freeswitch.org <mailto:brian at freeswitch.org>> wrote:
>>> 
>>> Sounds like there may be a bug in the yealink if packet size is of issue on TCP/TLS connections.
>>> 
>>> On Thu, Feb 25, 2016 at 5:13 PM, Emrah <lists at kavun.ch <mailto:lists at kavun.ch>> wrote:
>>> Hello list,
>>> I thought I had solved this issue by reducing my codec list  to a minimum, but it still persists, unfortunately. This was to reduce the TLS packet size.
>>> Anytime I enable SRTP on my phones, outgoing calls will randomly fail. The problem goes away when I disable SRTP. I only work over TLS.
>>> Incoming calls work reliably with or without SRTP.
>>> 
>>> How do you suggest debugging this?
>>> I tried setting up a fresh instance of FS but the issue persists.
>>> Now. It should be noted that calls fail sensibly more often when I have more than one account registered on the same server with the same device.
>>> 
>>> Any suggestion is welcome. Have you experienced this? 
>>> 
>>> I?m running FreeSWITCH Version 1.6.5 on Debian. My phone in this case is a Yealink SIP-T46G running firmware 28.80.0.95.
>>> 
>>> E
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>>> http://www.cluecon.com <http://www.cluecon.com/>
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>> 
>>> 
>>> 
>>> -- 
>>> Brian West
>>> brian at freeswitch.org <mailto:brian at freeswitch.org>
>>> 
>>> Twitter: @FreeSWITCH , @briankwest
>>> http://www.freeswitchbook.com <http://www.freeswitchbook.com/>
>>> http://www.freeswitchcookbook.com <http://www.freeswitchcookbook.com/>
>>> 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 <mailto:consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/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
>> 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  Sun Feb 28 21:37:45 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Sun, 28 Feb 2016 19:37:45 +0100
Subject: [Freeswitch-users] MOD_VERTO Documentation
In-Reply-To: <a2mogxx2kvmcxtd607rh3qbxf-0@mailer.nylas.com>
References: <CA+bkYdFUrCJyHn5FyYBmuvhiDDLPxv5GpBAxs5hzy4aernKOEA@mail.gmail.com>
	<a2mogxx2kvmcxtd607rh3qbxf-0@mailer.nylas.com>
Message-ID: <CAC7ttFEQNgemP+SAHUoahb2=TbRqgXR4hcrLnX0uGnZj6XF5AA@mail.gmail.com>

This tutorial is great! Thank you.

2016-02-28 18:14 GMT+01:00 ?talo Rossi <italo at freeswitch.org>:

> Hi Shlomi,
>
> We have a detailed doc here, but using the javascript lib, maybe it can
> give you some light, take a look at http://evoluxbr.github.io/verto-docs/
>
>> On Feb 28 2016, at 1:08 pm, Shlomi Schwartz <shlomis at liveperson.com>
>> wrote:
>> Hi all,
>>
>> Hello, I'm trying to implement mod_verto JSONRPC on IOS, is there any
>> documentation regarding the dialogParams on verto.invite method?
>>
>> The only documentation I found was :
>> https://freeswitch.org/confluence/display/FREESWITCH/mod_verto
>>
>> Thanks :)
>>
>> --
>> Shlomi Schwartz
>> R&D Technical Leader
>> T: +972-74-700-4511
>> <http://www.linkedin.com/company/164748> <http://twitter.com/liveperson>
>> <http://www.facebook.com/LivePersonInc> We Create Meaningful Connections
>>
>> <http://info.liveperson.com/201602/liz-welch/liveperson-robert-locascio-on-death-of-the-800-call.html>
>>
>> This message may contain confidential and/or privileged information.
>> If you are not the addressee or authorized to receive this on behalf of
>> the addressee you must not use, copy, disclose or take action based on this
>> message or any information herein.
>> If you have received this message in error, please advise the sender
>> immediately by reply email and delete this message. Thank you.
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
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 ssinyagin at gmail.com  Mon Feb 29 00:20:20 2016
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sun, 28 Feb 2016 22:20:20 +0100
Subject: [Freeswitch-users] Recording and mono directional webrtc video
In-Reply-To: <loom.20160228T115943-266@post.gmane.org>
References: <loom.20160222T182837-828@post.gmane.org>
	<loom.20160228T115943-266@post.gmane.org>
Message-ID: <CACOYK=j__m_M+bzMkMgo8xAmx=MTFJPgxteB_5weDAxdg0neCw@mail.gmail.com>

Doesn't seem impossible, but needs testing. If you have a budget, there are
some people who can help you build a prototype and troubleshoot the
problems if there are any.
On 28 Feb 2016 17:06, "Mario Giammarco" <mgiammarco at gmail.com> wrote:

> Mario Giammarco <mgiammarco at ...> writes:
>
> > I need a bidirectional audio
> > call with one direction video only from caller. And I need to record
> audio
> > and video from caller.
>
> Hello,
> nobody can help me? I am asking only if it is possible and supported.
>
> Thanks,
> Mario
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 freeswitch.opencode at spamgourmet.com  Mon Feb 29 01:43:05 2016
From: freeswitch.opencode at spamgourmet.com (freeswitch.opencode at spamgourmet.com)
Date: Sun, 28 Feb 2016 22:43:05 +0000
Subject: [Freeswitch-users] Getting 503 when starting automatically
In-Reply-To: <CAAkW=Ffrx-QY8N3O1Mz4uOPvetGA4BAWtXF-No41O5SGAK+KGw@mail.gmail.com>
References: <BN3PR13MB078550D45CAAEDE31D682A60D7A60@BN3PR13MB0785.namprd13.prod.outlook.com>,
	<CAAkW=Ffrx-QY8N3O1Mz4uOPvetGA4BAWtXF-No41O5SGAK+KGw@mail.gmail.com>
Message-ID: <BN3PR13MB07856D1CB9060AA6DBCE14E5D7B90@BN3PR13MB0785.namprd13.prod.outlook.com>

I enabled sofia tracing while the problem was occurring. I haven't had a chance to look into the source code yet, but so far I'm guessing it's not a permissions issue and something more like the invalid connection handle caching hypothesis.

Here's the sofia log snippet (DNS names and IP addresses sanitized):
nta.c:10803 outgoing_query_a() nta: for "myvoip.example.com" query "myvoip.example.com" A (cached)
nta.c:10856 outgoing_answer_a() nta: myvoip.example.com. IN A 10.0.0.1
tport.c:3257 tport_tsend() tport_tsend(0xb5b392e8) tpn = udp/10.0.0.1:5060
tport.c:4046 tport_resolve() tport_resolve addrinfo = 10.0.0.1:5060
tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0xb5b392e8): not found by name udp/10.0.0.1:5060
tport.c:3636 tport_send_fatal() tport_vsend(0xb5b392e8): Invalid argument with (s=22 udp/10.0.0.1:5060)
tport.c:3492 tport_send_msg() tport_vsend returned -1

David


From vishal.sharma at knowlarity.com  Mon Feb 29 07:54:13 2016
From: vishal.sharma at knowlarity.com (Vishal Sharma)
Date: Mon, 29 Feb 2016 10:24:13 +0530
Subject: [Freeswitch-users] File not found
Message-ID: <CAEaAqfjWwUzbZVB3Hp_P-Qmn3zRCRr9mLLkPOk6QyWc2MN5Ycg@mail.gmail.com>

Hi,
I am using FS 1.6.5, when FS tries to play a file
(/doesnotexist/srv/sounds/automation/media-sec.mp3) which doesn't exist on
file system, it takes 20 seconds for it to respond that file is not present
and dead air is played during this time.



On FS 1.4 and 1.2, it used to respond instantly.

Am i missing some setting ...

Regards,
Vishal Sharma

-- 
SuperReceptionist is now available on Android mobiles. Track your business 
on the go with call analytics, recordings, insights and more: Download the 
app here 
<https://play.google.com/store/apps/details?id=com.superreceptionist>
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From vishal.sharma at knowlarity.com  Mon Feb 29 08:50:59 2016
From: vishal.sharma at knowlarity.com (Vishal Sharma)
Date: Mon, 29 Feb 2016 11:20:59 +0530
Subject: [Freeswitch-users] Channel Hangup not received for Leg A
Message-ID: <CAEaAqfhsybg5TeoeStCk3M6kujEBwZGusCzRSMrp4W8w4v2nFw@mail.gmail.com>

I used following call string to bridge legB to legA

{ignore_early_media=true,originate_timeout=150,legA_UUID=36a59b97-32ea-42f5-9b4e-89e19a015f56,origination_caller_id_number=42994299,client_id_sys=1,client_meta_id_sys=None,ivr_refnum=820000222}
[leg_timeout=30]freetdm/outgoing/r/09506360596:_:{ignore_early_media=true,originate_timeout=150,legA_UUID=36a59b97-32ea-42f5-9b4e-89e19a015f56,origination_caller_id_number=42994299,client_id_sys=1,client_meta_id_sys=None,ivr_refnum=820000222}
[leg_timeout=30]freetdm/outgoing/r/08601224488


If legA disconnect before any of LegB is bridged, I get Cannel_hangup for
both legBs,  and channel_hangup_complete for legA but no channel_hangup for
legA.

I am using 1.6.5



Regards,
Vishal Sharma

-- 
SuperReceptionist is now available on Android mobiles. Track your business 
on the go with call analytics, recordings, insights and more: Download the 
app here 
<https://play.google.com/store/apps/details?id=com.superreceptionist>
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From andrew.keil at visytel.com  Mon Feb 29 09:37:02 2016
From: andrew.keil at visytel.com (Andrew Keil)
Date: Mon, 29 Feb 2016 06:37:02 +0000
Subject: [Freeswitch-users] Re- End Lua script after HangupHook handled
 without all the extra code to handle the return to the function
In-Reply-To: <CAFmZFiMFsg5JYGE1Ce6=CVdmiUY5+FUVbm5NHFKE_+enPFNC1g@mail.gmail.com>
References: <SY3PR01MB015646922642A69251933890F3A70@SY3PR01MB0156.ausprd01.prod.outlook.com>
	<CAFmZFiMFsg5JYGE1Ce6=CVdmiUY5+FUVbm5NHFKE_+enPFNC1g@mail.gmail.com>
Message-ID: <MEXPR01MB0151BDA1929ACA8A6E1F8ED8F3BA0@MEXPR01MB0151.ausprd01.prod.outlook.com>

Thanks for your response.

I have gone through these with no luck.   Like I said the session:destroy(???) crashes FreeSWITCH, which is therefore off the list.  The rest simply interrupt the current function and do no end the script.

I guess my next move is to see why session:destroy() crashes FreeSWITCH, however I am a little snowed under at the moment so if anyone has some time to replicate this (only needs one line of code in a Lua script) and pass this on to the developers that would be great.

Andrew

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Abaci B
Sent: Saturday, 27 February 2016 1:29 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Re- End Lua script after HangupHook handled without all the extra code to handle the return to the function

See https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:setHangupHook for a few ways to exit the lua script (error(), return "exit", return "die", s:destroy("error message")). I personally tried return "exit" but it seems to me that it only exits the calling function, haven't had a chance to look further, it's possible that the calling it from the within a function is different. if you play around and figure out please report back.

On Thu, Feb 25, 2016 at 9:33 PM, Andrew Keil <andrew.keil at visytel.com<mailto:andrew.keil at visytel.com>> wrote:
To FreeSWITCH Users,

See below for a sample template for a Lua Service Script running inside FreeSWITCH.

The issue I have is fairly straightforward.

I need a function to run when hangup is detected (ie. at the end of the call) however I understand this must not delay ending the script.  This function is CleanUp().  Then I would like the service to end.

The problem I am having is if the caller hangs up during the playback of ?intro.wav? (as shown inside the MainService() function below), then the code jumps to the myHangupHook which calls CleanUp() perfectly, the issue is once CleanUp() is complete I would like the Lua script to end there and then (ie. at the bottom of CleanUp()).  What actually happens is it returns to MainService() and continues to try and play ?info.wav?, unless I either check for session:ready() everywhere or add a goto as shown below under each streamFile() function call.

My aim is to reduce extra code and to make the Lua script simpler and easier to read.  Also I would like to try and avoid goto statements, which I know can be done with if (session:ready()) etc?.

So is there a way to stop a Lua script running inside FreeSWITCH cleanly?  I have tried the os.exit() this is barred from use by FreeSWITCH.  I have also tried session:destroy() which crashes FreeSWITCH (version 1.6.5 on CentOS 6.7, CentOS 7 and windows) 100% of the time!

I could look further into the Lua additions done by the FreeSWITCH team in the source code, however if someone has already solved this then that would be the best solution.

FYI: Obviously the script below is simple, however I am sure that you understand if the script was complicated having to use ?if (session:ready()) then ?.? or ?if (not session:ready()) then goto HANGUPEXIT end? makes the code ugly.

Thanks in advance,

Andrew Keil
Visytel Pty Ltd



------------------------------------------------------------------------------  Sample Lua Service -----------------------------------------------------------------------

-- Lua template for FreeSWITCH service
-- By: Andrew Keil (Visytel Pty Ltd)
-- Email: support at visytel.com<mailto:support at visytel.com>

-- Setup script wide variables here

function PreAnswer()
                freeswitch.consoleLog("INFO", "PRE ANSWER SECTION\n");
                -- Add your pre answer code from here

                -- End of your pre answer code
                freeswitch.consoleLog("INFO", "PRE ANSWER SECTION COMPLETE\n");
end

function AnswerCaller()
                session:answer()
                session:sleep(1000)
end

function MainService()
                freeswitch.consoleLog("INFO", "MAIN SERVICE SECTION\n");
                if (session:ready()) then
                                -- Note (1): If you wish to end the call then simply use: goto ENDSERVICE
                                -- Note (2): To terminate the service sooner when HANGUP is detected use: if (not session:ready()) then goto HANGUPEXIT end
                                -- Add your main service code from here               (caller would have been answered)

                                session:streamFile("intro.wav")
                                if (not session:ready()) then goto HANGUPEXIT end
                                session:streamFile("info.wav")
                                if (not session:ready()) then goto HANGUPEXIT end
                                session:streamFile("outro.wav")
                                if (not session:ready()) then goto HANGUPEXIT end

                                -- End of your main service code
                end
                ::ENDSERVICE::
                if (session:ready()) then
                                -- End of service so hangup
                                session:hangup()  -- Should automatically jump to CleanUp() via hangup handler if caller still online at this stage
                end
                goto END
                ::HANGUPEXIT::
                freeswitch.consoleLog("INFO", "END OF SERVICE (HANGUP DETECTED)\n");
                ::END::
                freeswitch.consoleLog("INFO", "MAIN SERVICE SECTION COMPLETE\n");
end

function CleanUp()
                freeswitch.consoleLog("INFO", "CLEANUP SECTION\n");
                -- Add your cleanup code from here (caller would have been disconnected)

                -- End of your cleanup code
                freeswitch.consoleLog("INFO", "CLEANUP SECTION COMPLETE\n");
end

function myHangupHook(s, status, arg)
                session:hangup()
                CleanUp() -- Run CleanUp function now since the caller has disconnected
end

-- Setup Hangup event handler here
v_hangup = "HANGUP"
session:setHangupHook("myHangupHook", "v_hangup")

-- Call service functions in order
PreAnswer()
AnswerCaller()
MainService()
-- End of Lua service



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

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

FreeSWITCH-users mailing list
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From Adam.Seeliger at qsc.de  Mon Feb 29 11:29:11 2016
From: Adam.Seeliger at qsc.de (Seeliger, Adam)
Date: Mon, 29 Feb 2016 08:29:11 +0000
Subject: [Freeswitch-users] all-reg-options-ping and tls issue
In-Reply-To: <CAHtxdDfuv86b8a0EsG=6yW3QD1KwXbmtNs5_7O6SnpbX1PFJ=g@mail.gmail.com>
References: <2E67ADAE1D3582409C90EBAE8C64C5070149F6B3@QSCDEMXP01a.ONE4ALL.LAN>
	<6D3DEA47-DECB-4509-A327-B8DFEC1C1A83@kavun.ch>
	<2E67ADAE1D3582409C90EBAE8C64C507122D35C1@QSCDEMXP01a.ONE4ALL.LAN>
	<CAHtxdDdy-nq_gZDLKW5bE62-5TQV2u-fK=sSmVD5Y07qv=0Cxw@mail.gmail.com>
	<2E67ADAE1D3582409C90EBAE8C64C507122D35E9@QSCDEMXP01a.ONE4ALL.LAN>
	<CAHtxdDeAcZis=3SZnVwoK9j=29SLPTobDQUFGKSjLPfcFXGFTg@mail.gmail.com>
	<2E67ADAE1D3582409C90EBAE8C64C507122D361C@QSCDEMXP01a.ONE4ALL.LAN>
	<CAHtxdDfuv86b8a0EsG=6yW3QD1KwXbmtNs5_7O6SnpbX1PFJ=g@mail.gmail.com>
Message-ID: <2E67ADAE1D3582409C90EBAE8C64C507122D372E@QSCDEMXP01a.ONE4ALL.LAN>

Hi,

I also tested it on latest master:
FreeSWITCH Version 1.7.0+git~20160227T004333Z~d89a0ad52d~64bit (git d89a0ad 2016-02-27 00:43:33Z 64bit)

FreeSWITCH still does not send OPTIONs to TLS registered Users and kills their Registration ?

Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Sergey Safarov
Gesendet: Freitag, 26. Februar 2016 16:53
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] all-reg-options-ping and tls issue


Registration is correct.
Think it bug and requred to fill a jira tiket.
I use master 4 mouth old and it work correctly.

On Fri, Feb 26, 2016, 18:00 Seeliger, Adam <Adam.Seeliger at qsc.de<mailto:Adam.Seeliger at qsc.de>> wrote:
Hi, here is the requested output.

I changed the real user, domain and ip address values into descriptions.
I guess the values are not necessary?

Both, FreeSWITCH and the phone are in the same network (no NAT involved here)

Registrations:
=================================================================================================
Call-ID:         3134353634383833373032323631-ncwgvit2obfp
User:            user at domain
Contact:       "User Name" <sip:user at ip:60206;transport=tls;line=gvg9q8jh;fs_nat=yes;fs_path=sip%3Auser%40ip%3A60206>
Agent:          snom715/8.7.5.35<http://8.7.5.35>
Status:         Registered(AUTO-NAT-2.0)(unknown) EXP(2016-02-26 16:21:32) EXPSECS(1912)
Ping-Status:   Reachable
Host:            hostname
IP:               ip
Port:            60206
Auth-User:    user
Auth-Realm: domain
MWI-Account:          user at domain

Total items returned: 1
=================================================================================================
2016-02-26 15:50:09.326648 [WARNING] sofia.c:5769 Sip user 'user at domain' is now Unreachable
2016-02-26 15:50:09.326648 [WARNING] sofia.c:5780 Expire sip user 'user at domain' due to options failure

As you can see, the user immediately got unregistered again ?

Best regards,
Adam

Von: 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>] Im Auftrag von Sergey Safarov
Gesendet: Freitag, 26. Februar 2016 15:45

An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] all-reg-options-ping and tls issue

Please send output of command "sofia status profile internal reg <exten_number>"

On Fri, Feb 26, 2016 at 4:21 PM, Seeliger, Adam <Adam.Seeliger at qsc.de<mailto:Adam.Seeliger at qsc.de>> wrote:
Hi,

the phone uses sip+tls.
I test using a snom715, got plenty other phones here, but I guess they will behave the same way.
It really looks like FreeSWITCH is doing something wrong (or is wrongly configured ? if there are any parameters for options ping supporting both, udp and tls)

Regards

Von: 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>] Im Auftrag von Sergey Safarov
Gesendet: Freitag, 26. Februar 2016 13:59

An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] all-reg-options-ping and tls issue

If your phone has enabled SIPS uri please disable and use sip+tls.


On Fri, Feb 26, 2016 at 3:37 PM, Seeliger, Adam <Adam.Seeliger at qsc.de<mailto:Adam.Seeliger at qsc.de>> wrote:
Hi and thanks for the feedback,

sry that I did not respond for a long time.

I already use:
<variable name=?sip-force-contact? value="NDLB-connectile-dysfunction-2.0"/>

I also tested all mentioned params below, nothing works.

When I register a User via TLS FreeSWITCH does not even try to ping the user.
I turned sofia global siptrace on and watched the flow:

User                                        Server
13:09:33.311446:             REGISTER [TLS] ->
13:09:33.312552:             <- 401 UNAUTHORIZED [TLS]
13:09:33.331948:             REGISTER (AUTH) [TLS]  ->
13:09:33.336959:             <- 200 OK [TLS]
Nothing happens
2016-02-26 13:10:00.619525 [WARNING] sofia.c:5769 Sip user 'user at host' is now Unreachable
2016-02-26 13:10:00.619525 [WARNING] sofia.c:5780 Expire sip user 'user at host' due to options failure

When I REGISTER the User via UDP FreeSWITCH starts to ping (OPTIONS) the user as soon as he is registered.

Is there any way to force FreeSWITCH to send OPTIONs in both, udp and tls, depending on the registration?

Thanks in advance,
Adam

Von: 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>] Im Auftrag von Emrah
Gesendet: Freitag, 29. Januar 2016 09:25
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] all-reg-options-ping and tls issue

Hi!
This is interesting. I experienced something rather similar where calls would drop because FS would timeout on certain packets sent over UDP instead of TLS.
I assume you mean FS exits with port 5060 instead of port 5061? Because the port on the remote end should be dynamically set.
I found out that in my case, what works best even with TLS, is to use:
<variable name=?sip-force-contact? value="NDLB-connectile-dysfunction-2.0"/>
This goes as far as it can to lay out the path to contacting the client with all consideration in regards to NAT and dynamic ports.
Not sure if it will help you. I?ve personally disabled options-ping an let my clients deal with keep-alive instead.

You could also look into:
    <!-- add a ;received="<ip>:<port>" to the contact when replying to register for nat handling -->
<!? <param name="NDLB-received-in-nat-reg-contact" value="true"/> -->
<!? <param name="bind-params" value="transport=tcp,udp,tls"/>  -->
<!-- additional bind parameters for TLS -->
    <param name=?tls-bind-params" value="transport=tls"/>

I?ll leave it up to you to investigate those options more in details on the FS documentation.

Please keep us posted!

E
On Jan 28, 2016, at 11:48 AM, Seeliger, Adam <Adam.Seeliger at qsc.de<mailto:Adam.Seeliger at qsc.de>> wrote:

Hi all,

I have a problem, when I enable TLS and register a phone using TLS on Port 5061.
FreeSWITCH still tries to ?ping? the phone using Port 5060 using UDP, which is ignored by the phone.
Moments later FreeSWITCH deletes the registration, because ?unregister-on-options-fail? is set to ?true?.

I already figured out, that you can set ?all-reg-options-ping? to ?udp-only?, but this would completely disable this feature for TLS.
Is there any way to ping TLS registered using TLS?

Thanks in advance

- Adam

_________________________________________________________________________
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
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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<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 dominique.jeannerod at interact-iv.com  Mon Feb 29 11:52:12 2016
From: dominique.jeannerod at interact-iv.com (Dominique Jeannerod)
Date: Mon, 29 Feb 2016 09:52:12 +0100
Subject: [Freeswitch-users] mod_distributor / gwlist down
In-Reply-To: <CAEzwe14uLCk+MvyQ8mNuixR1Gy13D8ox+TKbOTt9ucQEWdePsw@mail.gmail.com>
References: <CAEzwe14uLCk+MvyQ8mNuixR1Gy13D8ox+TKbOTt9ucQEWdePsw@mail.gmail.com>
Message-ID: <CAEzwe17VkJ_iHKvHHPfSLMK8GMqeP2XTxKinqYn=usJxdKgggw@mail.gmail.com>

Hello,

I worked again on this, and there is no problem at all related with
mod_distributor.
What I found : there is a problem evaluating nested expressions.
This code works as expected :
    <action application="set" data="gwlist_down=${sofia profile
internal_trk_1 gwlist down}"/>
    <action application="set" data="gateway=${distributor I_MUT_SIPGTW
 ${gwlist_down}}"/>
    <action application="bridge"
data="sofia/gateway/${gateway}/${destination_number}"/>

This code is not OK :
    <action application="bridge" data="sofia/gateway/${distributor
I_MUT_SIPGTW ${sofia internal_trk_1 gwlist down}}/${destination_number}"/>
?
Any idea on the right syntax to make it work on one line as it should ?


Best regards

Dominique Jeannerod
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From idokan at gmail.com  Mon Feb 29 15:59:47 2016
From: idokan at gmail.com (ik)
Date: Mon, 29 Feb 2016 14:59:47 +0200
Subject: [Freeswitch-users] Debugging Perl code
In-Reply-To: <CAOccsKDLPHDkoqV2RLEmfnNAbRQ=BGtKwP_V3sL6W1QeG87qvQ@mail.gmail.com>
References: <CAOccsKDLPHDkoqV2RLEmfnNAbRQ=BGtKwP_V3sL6W1QeG87qvQ@mail.gmail.com>
Message-ID: <CAOccsKCU+1hZtWGtNhGDDd0f2ScQwohAjMacvT5HuAOvgZic4w@mail.gmail.com>

Hello,

I have Perl code that does new freeswitch::Session.

When that happens, freeswitch crashes.

It's an old freeswitch v1.4.26.

I execute it from now using the cli either by perl <file.pl> or perlrun.

How can I debug the cause of such crash?

Thank you
Ido
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From rutu.patel at inextrix.com  Mon Feb 29 16:15:28 2016
From: rutu.patel at inextrix.com (Rutu Patel)
Date: Mon, 29 Feb 2016 18:45:28 +0530
Subject: [Freeswitch-users] Call dropping after 32 seconds
In-Reply-To: <CAGYpU64NgPRKPCywdNn3whBh8VH6=eYLy=PJPGrYn8JJ5iMPGQ@mail.gmail.com>
References: <CAPQet1zs08bGNnGDVmsLH3R3VbvMCJW=8Cx3U+KPxwGpYSuzAA@mail.gmail.com>
	<0000015307b150a5-132a41d6-5bea-4af1-9686-2653df68a3b6-000000@email.amazonses.com>
	<CAPQet1wJPNZj7S4ezzd=OSDqHzLKQMieOWvtL8Qm5ht1PcHyBw@mail.gmail.com>
	<CAGYpU64NgPRKPCywdNn3whBh8VH6=eYLy=PJPGrYn8JJ5iMPGQ@mail.gmail.com>
Message-ID: <CAPQet1xzzXKUk=4jJRqtZ4rwtLUDEtr7n-eZBGvQp6kvinb+aw@mail.gmail.com>

Hi Jurijs,

We have to consider the setup with freeswitch and opensips only.
registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips
server)

And there is a firewall and we have added x.x.x.3 IP there.

Is it possible there is some network related issue Or any sip profile
parameter we need to set?

--
Thanks,
Rutu Patel

On Tue, Feb 23, 2016 at 1:07 PM, Jurijs Ivolga <jurij.ivo at gmail.com> wrote:

> Hi,
>
> 1) You have very complex set-up and I doubt that you need it.
>
> 2) As far as you have user with ip x.x.x.174 and opensips server with same
> ip x.x.x.174 it very hard to debug. So I propose you to send new log where
> will be difference between user ip and opensips IP.
>
> 3) If you have possibility, try to register directly with a user to
> x.x.x.3 gateway and check if same issue still exists, if there is no such
> issue anymore, then thee is definitely issue in your opensips x.x.x.174 and
> freeswitch x.x.x.166. My point here is that you need to isolate issue and
> to understand what part of your set-up works as expected and what is faulty.
>
> With kind regards,
>
> Jurijs
>
> 2016-02-23 7:51 GMT+02:00 Rutu Patel <rutu.patel at inextrix.com>:
>
>> Thanks for the reply.
>>
>> Got your point about NATing issue and no response of 200 OK and as a
>> resoult ACK Timeout.
>> So, now to resolve the issue, if you can assist, what could be the
>> possible fixies?
>> From where can i start? where to look?
>>
>> Thanks.
>>
>> --
>> Thanks,
>> Rutu Patel
>>
>> <http://www.inextrix.com>
>>
>> On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> 5 second response: 32 seconds is a timer/[network/NAT] issue.
>>>
>>> You have lots of 200s to the user since it's waiting for an ACK and
>>> keeps retrying, but for whatever network reason (router... sip alg?), it
>>> isn't getting one, so it triggers a timer to stop the call.
>>>
>>>
>>> -Avi Marcus
>>> BestFone
>>>
>>> On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel <rutu.patel at inextrix.com>
>>> wrote:
>>>
>>>> Hello,
>>>>
>>>> Having issue of call dropping after 32 seconds, here are the details-
>>>>
>>>> x.x.x.174: opensips server
>>>> x.x.x.166: freeswitch server
>>>> x.x.x.3:     another opensips server which is registered as gateway on
>>>> above freeswitch server
>>>> x.x.x.6: freeswitch server
>>>> x.x.x.47:  server through which the user is registered
>>>> I am trying to call from xxxx9 to xxxxxxx29858
>>>> xxxxxxx00181 is caller-id name and caller-id number
>>>>
>>>> Call flow is like this:
>>>> registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips
>>>> server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server)
>>>>
>>>> Outbound-proxy is set to x.x.x.174 in Gateway configuration.
>>>>
>>>> 1) Call is initiated by the user(xxxx9) registered with host x.x.x.174
>>>> 2) Call hit the freeswitch server x.x.x.166
>>>> 3) After '180 Ringing' and '183 Session Progress' packet
>>>> sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the
>>>> gateway x.x.x.3
>>>> But after 32 seconds call is dropped,
>>>> Within 32 seconds audio is ok from both end so it should not be the RTP
>>>> issue.
>>>> Here I have attached the file with sip logs, you can observer from the
>>>> file that, there are many '200 OK' from x.x.x.174 to x.x.x.166
>>>> and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then
>>>> 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped.
>>>>
>>>> What is wrong here? Any help would be appreciated here.
>>>>
>>>> Here is the file with sip logs
>>>> --
>>>> Thanks,
>>>> Rutu Patel
>>>> <http://www.inextrix.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
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> Jurijs
>
> 2016-02-23 7:51 GMT+02:00 Rutu Patel <rutu.patel at inextrix.com>:
>
>> Thanks for the reply.
>>
>> Got your point about NATing issue and no response of 200 OK and as a
>> resoult ACK Timeout.
>> So, now to resolve the issue, if you can assist, what could be the
>> possible fixies?
>> From where can i start? where to look?
>>
>> Thanks.
>>
>> --
>> Thanks,
>> Rutu Patel
>>
>> <http://www.inextrix.com>
>>
>> On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> 5 second response: 32 seconds is a timer/[network/NAT] issue.
>>>
>>> You have lots of 200s to the user since it's waiting for an ACK and
>>> keeps retrying, but for whatever network reason (router... sip alg?), it
>>> isn't getting one, so it triggers a timer to stop the call.
>>>
>>>
>>> -Avi Marcus
>>> BestFone
>>>
>>> On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel <rutu.patel at inextrix.com>
>>> wrote:
>>>
>>>> Hello,
>>>>
>>>> Having issue of call dropping after 32 seconds, here are the details-
>>>>
>>>> x.x.x.174: opensips server
>>>> x.x.x.166: freeswitch server
>>>> x.x.x.3:     another opensips server which is registered as gateway on
>>>> above freeswitch server
>>>> x.x.x.6: freeswitch server
>>>> x.x.x.47:  server through which the user is registered
>>>> I am trying to call from xxxx9 to xxxxxxx29858
>>>> xxxxxxx00181 is caller-id name and caller-id number
>>>>
>>>> Call flow is like this:
>>>> registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips
>>>> server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server)
>>>>
>>>> Outbound-proxy is set to x.x.x.174 in Gateway configuration.
>>>>
>>>> 1) Call is initiated by the user(xxxx9) registered with host x.x.x.174
>>>> 2) Call hit the freeswitch server x.x.x.166
>>>> 3) After '180 Ringing' and '183 Session Progress' packet
>>>> sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the
>>>> gateway x.x.x.3
>>>> But after 32 seconds call is dropped,
>>>> Within 32 seconds audio is ok from both end so it should not be the RTP
>>>> issue.
>>>> Here I have attached the file with sip logs, you can observer from the
>>>> file that, there are many '200 OK' from x.x.x.174 to x.x.x.166
>>>> and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then
>>>> 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped.
>>>>
>>>> What is wrong here? Any help would be appreciated here.
>>>>
>>>> Here is the file with sip logs
>>>> --
>>>> Thanks,
>>>> Rutu Patel
>>>> <http://www.inextrix.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
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 ynasida at gmail.com  Mon Feb 29 16:20:31 2016
From: ynasida at gmail.com (=?UTF-8?B?0K7RgNC40Lkg0J3QsNGB0LjQtNCw?=)
Date: Mon, 29 Feb 2016 16:20:31 +0300
Subject: [Freeswitch-users] where mod_com_g729 ?
In-Reply-To: <DE690397-43CB-420F-BAC5-5E7DA57B81E3@freeswitch.org>
References: <CADihe3J3kv2RJ8DeueusW6aAkzGWNxR2h5zaKiLKV9asJg6LVg@mail.gmail.com>
	<CABMX_w-ga8GM9s+JHyWjSOAMLRqfJ=o5+YTLYww5V8qNbD7A4Q@mail.gmail.com>
	<CADihe3JFLUtzLPQk6J+0rsVtVP5FG88uhbjyNeBOfhqP1brWfg@mail.gmail.com>
	<DE690397-43CB-420F-BAC5-5E7DA57B81E3@freeswitch.org>
Message-ID: <CADihe3+3xCRQXtoKJsy4=Oc6XaZXHCYEqfxME8na_Oh=s_TXhA@mail.gmail.com>

Thanks guys.
I just had to set manually the path for  script fs-latest-installer-v1.6
<bin_path> <mods_path> <config_path>
Otherwise mod_com_g729.so was not installed anywhere (without any warning
message).

2016-02-28 0:00 GMT+03:00 Ken Rice <krice at freeswitch.org>:

> I'm pretty sure the link he gave you is the g729 commercial mod
> installation instructions and answers your question
>
> Sent from my iPhone
>
> On Feb 27, 2016, at 12:57 PM, ???? ?????? <ynasida at gmail.com> wrote:
>
> Looks like you didn't read my question.
>
> 2016-02-27 18:49 GMT+03:00 ?talo Rossi <italo at freeswitch.org>:
>
>> https://freeswitch.org/confluence/display/FREESWITCH/mod_com_g729
>>
>> On Sat, Feb 27, 2016 at 5:36 AM, ???? ?????? <ynasida at gmail.com> wrote:
>>
>>> Hi list
>>>
>>> I have installed FS 1.6.6 via apt-get method (all modules) and don't see
>>> mod_com_g729 anywhere.
>>>
>>> Should I install it from source ?
>>>
>>> Please advice.
>>>
>>> # apt-cache search freeswitch | grep 729
>>> freeswitch-mod-g729 - mod_g729 for FreeSWITCH
>>> freeswitch-mod-g729-dbg - mod_g729 for FreeSWITCH (debug)
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 shlomis at liveperson.com  Mon Feb 29 16:43:16 2016
From: shlomis at liveperson.com (Shlomi Schwartz)
Date: Mon, 29 Feb 2016 15:43:16 +0200
Subject: [Freeswitch-users] MOD_VERTO Documentation
In-Reply-To: <CA+bkYdFUrCJyHn5FyYBmuvhiDDLPxv5GpBAxs5hzy4aernKOEA@mail.gmail.com>
References: <CA+bkYdFUrCJyHn5FyYBmuvhiDDLPxv5GpBAxs5hzy4aernKOEA@mail.gmail.com>
Message-ID: <CA+bkYdHw0YJn1Ap-89EBJiZROypbJParF7RROdRMStKAKgP+Zg@mail.gmail.com>

super, thanks

On Sun, Feb 28, 2016 at 5:17 PM, Shlomi Schwartz <shlomis at liveperson.com>
wrote:

> Hi all,
>
> Hello, I'm trying to implement mod_verto JSONRPC on IOS, is there any
> documentation regarding the dialogParams on verto.invite method?
>
> The only documentation I found was :
> https://freeswitch.org/confluence/display/FREESWITCH/mod_verto
>
> Thanks :)
>
> --
> Shlomi Schwartz
> R&D Technical Leader
> T: +972-74-700-4511
> <http://www.linkedin.com/company/164748> <http://twitter.com/liveperson>
> <http://www.facebook.com/LivePersonInc> We Create Meaningful Connections
>
> <http://info.liveperson.com/201602/liz-welch/liveperson-robert-locascio-on-death-of-the-800-call.html>
>



-- 
Shlomi Schwartz
R&D Technical Leader
T: +972-74-700-4511
<http://www.linkedin.com/company/164748> <http://twitter.com/liveperson>
<http://www.facebook.com/LivePersonInc> We Create Meaningful Connections
<http://info.liveperson.com/201602/liz-welch/liveperson-robert-locascio-on-death-of-the-800-call.html>

-- 
This message may contain confidential and/or privileged information. 
If you are not the addressee or authorized to receive this on behalf of the 
addressee you must not use, copy, disclose or take action based on this 
message or any information herein. 
If you have received this message in error, please advise the sender 
immediately by reply email and delete this message. Thank you.
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From ahabiba at gmail.com  Mon Feb 29 17:13:06 2016
From: ahabiba at gmail.com (Ahmed Habiba)
Date: Mon, 29 Feb 2016 17:13:06 +0300
Subject: [Freeswitch-users] Call dropping after 32 seconds
In-Reply-To: <mailman.2775.1456751789.678.freeswitch-users@lists.freeswitch.org>
References: <mailman.2775.1456751789.678.freeswitch-users@lists.freeswitch.org>
Message-ID: <EC8A5E4B-87F9-43AB-A7D1-14773EDA1E6F@gmail.com>

Hi,

Where is the firewall, is it between user and freeswitch or freeswitch and opensips

All cases freeswitch should send the firewall IP for the endpoint behind firewall so that the endpoint would be able to communicate back to freeswitch correct external IP.

Thanks,

Ahmed Habiba.


From: Rutu Patel <rutu.patel at inextrix.com <mailto:rutu.patel at inextrix.com>>
Subject: Re: [Freeswitch-users] Call dropping after 32 seconds
Date: February 29, 2016 at 4:15:28 PM GMT+3
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>


Hi Jurijs,

We have to consider the setup with freeswitch and opensips only.
registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips server) 

And there is a firewall and we have added x.x.x.3 IP there.

Is it possible there is some network related issue Or any sip profile parameter we need to set?

--
Thanks,
Rutu Patel

On Tue, Feb 23, 2016 at 1:07 PM, Jurijs Ivolga <jurij.ivo at gmail.com <mailto:jurij.ivo at gmail.com>> wrote:
Hi,

1) You have very complex set-up and I doubt that you need it.

2) As far as you have user with ip x.x.x.174 and opensips server with same ip x.x.x.174 it very hard to debug. So I propose you to send new log where will be difference between user ip and opensips IP.

3) If you have possibility, try to register directly with a user to x.x.x.3 gateway and check if same issue still exists, if there is no such issue anymore, then thee is definitely issue in your opensips x.x.x.174 and freeswitch x.x.x.166. My point here is that you need to isolate issue and to understand what part of your set-up works as expected and what is faulty.

With kind regards,  

Jurijs

2016-02-23 7:51 GMT+02:00 Rutu Patel <rutu.patel at inextrix.com <mailto:rutu.patel at inextrix.com>>:
Thanks for the reply.

Got your point about NATing issue and no response of 200 OK and as a resoult ACK Timeout.
So, now to resolve the issue, if you can assist, what could be the possible fixies?
From where can i start? where to look?

Thanks.

--
Thanks,
Rutu Patel

 <http://www.inextrix.com/>
On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus <avi at avimarcus.net <mailto:avi at avimarcus.net>> wrote:
5 second response: 32 seconds is a timer/[network/NAT] issue.

You have lots of 200s to the user since it's waiting for an ACK and keeps retrying, but for whatever network reason (router... sip alg?), it isn't getting one, so it triggers a timer to stop the call.


-Avi Marcus
BestFone

On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel <rutu.patel at inextrix.com <mailto:rutu.patel at inextrix.com>> wrote:
Hello, 

Having issue of call dropping after 32 seconds, here are the details- 

x.x.x.174: opensips server 
x.x.x.166: freeswitch server 
x.x.x.3:     another opensips server which is registered as gateway on above freeswitch server
x.x.x.6: freeswitch server
x.x.x.47:  server through which the user is registered 
I am trying to call from xxxx9 to xxxxxxx29858 
xxxxxxx00181 is caller-id name and caller-id number 

Call flow is like this: 
registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server)

Outbound-proxy is set to x.x.x.174 in Gateway configuration.

1) Call is initiated by the user(xxxx9) registered with host x.x.x.174 
2) Call hit the freeswitch server x.x.x.166 
3) After '180 Ringing' and '183 Session Progress' packet sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the gateway x.x.x.3 
But after 32 seconds call is dropped, 
Within 32 seconds audio is ok from both end so it should not be the RTP issue. 
Here I have attached the file with sip logs, you can observer from the file that, there are many '200 OK' from x.x.x.174 to x.x.x.166 
and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped. 

What is wrong here? Any help would be appreciated here. 

Here is the file with sip logs
--
Thanks,
Rutu Patel
 <http://www.inextrix.com/>
_________________________________________________________________________
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
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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>
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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/>


Jurijs

2016-02-23 7:51 GMT+02:00 Rutu Patel <rutu.patel at inextrix.com <mailto:rutu.patel at inextrix.com>>:
Thanks for the reply.

Got your point about NATing issue and no response of 200 OK and as a resoult ACK Timeout.
So, now to resolve the issue, if you can assist, what could be the possible fixies?
From where can i start? where to look?

Thanks.

--
Thanks,
Rutu Patel

 <http://www.inextrix.com/>
On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus <avi at avimarcus.net <mailto:avi at avimarcus.net>> wrote:
5 second response: 32 seconds is a timer/[network/NAT] issue.

You have lots of 200s to the user since it's waiting for an ACK and keeps retrying, but for whatever network reason (router... sip alg?), it isn't getting one, so it triggers a timer to stop the call.


-Avi Marcus
BestFone

On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel <rutu.patel at inextrix.com <mailto:rutu.patel at inextrix.com>> wrote:
Hello, 

Having issue of call dropping after 32 seconds, here are the details- 

x.x.x.174: opensips server 
x.x.x.166: freeswitch server 
x.x.x.3:     another opensips server which is registered as gateway on above freeswitch server
x.x.x.6: freeswitch server
x.x.x.47:  server through which the user is registered 
I am trying to call from xxxx9 to xxxxxxx29858 
xxxxxxx00181 is caller-id name and caller-id number 

Call flow is like this: 
registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server)

Outbound-proxy is set to x.x.x.174 in Gateway configuration.

1) Call is initiated by the user(xxxx9) registered with host x.x.x.174 
2) Call hit the freeswitch server x.x.x.166 
3) After '180 Ringing' and '183 Session Progress' packet sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the gateway x.x.x.3 
But after 32 seconds call is dropped, 
Within 32 seconds audio is ok from both end so it should not be the RTP issue. 
Here I have attached the file with sip logs, you can observer from the file that, there are many '200 OK' from x.x.x.174 to x.x.x.166 
and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped. 

What is wrong here? Any help would be appreciated here. 

Here is the file with sip logs
--
Thanks,
Rutu Patel
 <http://www.inextrix.com/>
_________________________________________________________________________
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>
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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>
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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>
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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
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From brian at freeswitch.org  Mon Feb 29 17:30:31 2016
From: brian at freeswitch.org (Brian West)
Date: Mon, 29 Feb 2016 08:30:31 -0600
Subject: [Freeswitch-users] where mod_com_g729 ?
In-Reply-To: <CADihe3+3xCRQXtoKJsy4=Oc6XaZXHCYEqfxME8na_Oh=s_TXhA@mail.gmail.com>
References: <CADihe3J3kv2RJ8DeueusW6aAkzGWNxR2h5zaKiLKV9asJg6LVg@mail.gmail.com>
	<CABMX_w-ga8GM9s+JHyWjSOAMLRqfJ=o5+YTLYww5V8qNbD7A4Q@mail.gmail.com>
	<CADihe3JFLUtzLPQk6J+0rsVtVP5FG88uhbjyNeBOfhqP1brWfg@mail.gmail.com>
	<DE690397-43CB-420F-BAC5-5E7DA57B81E3@freeswitch.org>
	<CADihe3+3xCRQXtoKJsy4=Oc6XaZXHCYEqfxME8na_Oh=s_TXhA@mail.gmail.com>
Message-ID: <CAEJMVkCASCL46CYtuo0gH_gFDrV2bUCA3Tnd+qUs1Vz40PWvAA@mail.gmail.com>

Are you using a nonstandard install path?

On Monday, February 29, 2016, ???? ?????? <ynasida at gmail.com> wrote:

> Thanks guys.
> I just had to set manually the path for  script fs-latest-installer-v1.6
> <bin_path> <mods_path> <config_path>
> Otherwise mod_com_g729.so was not installed anywhere (without any warning
> message).
>
> 2016-02-28 0:00 GMT+03:00 Ken Rice <krice at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','krice at freeswitch.org');>>:
>
>> I'm pretty sure the link he gave you is the g729 commercial mod
>> installation instructions and answers your question
>>
>> Sent from my iPhone
>>
>> On Feb 27, 2016, at 12:57 PM, ???? ?????? <ynasida at gmail.com
>> <javascript:_e(%7B%7D,'cvml','ynasida at gmail.com');>> wrote:
>>
>> Looks like you didn't read my question.
>>
>> 2016-02-27 18:49 GMT+03:00 ?talo Rossi <italo at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','italo at freeswitch.org');>>:
>>
>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_com_g729
>>>
>>> On Sat, Feb 27, 2016 at 5:36 AM, ???? ?????? <ynasida at gmail.com
>>> <javascript:_e(%7B%7D,'cvml','ynasida at gmail.com');>> wrote:
>>>
>>>> Hi list
>>>>
>>>> I have installed FS 1.6.6 via apt-get method (all modules) and don't
>>>> see mod_com_g729 anywhere.
>>>>
>>>> Should I install it from source ?
>>>>
>>>> Please advice.
>>>>
>>>> # apt-cache search freeswitch | grep 729
>>>> freeswitch-mod-g729 - mod_g729 for FreeSWITCH
>>>> freeswitch-mod-g729-dbg - mod_g729 for FreeSWITCH (debug)
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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
>>>>
>>>
>>>
>>>
>>> --
>>> ?talo Rossi
>>> italo at freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','italo at 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


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

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 govoiper at gmail.com  Mon Feb 29 17:41:23 2016
From: govoiper at gmail.com (SamyGo)
Date: Mon, 29 Feb 2016 09:41:23 -0500
Subject: [Freeswitch-users] ESL Client --> Multiple FS Servers
Message-ID: <CAJUJwtjvsGr0smS-hAHu3KV3Cx3DgpqWZQpM7XVAbU0gSsaVRQ@mail.gmail.com>

Hi All,

I'm looking for some guidance on connecting my ESL client program to
multiple FS Servers at once. I've written a small code which connects to
one FS ESL and based on events it generates some reports and also send back
some actions to FS.

Now I'm looking to expand that form one to multiple FS servers but the code
should collect all events from every FS and be able to send action to any
of them whatever the logic be.

Any guides or hints will be much appreciated.

Regards,
Sammy
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From deforceczt at gmail.com  Mon Feb 29 18:23:48 2016
From: deforceczt at gmail.com (Vladislav Ivanov)
Date: Mon, 29 Feb 2016 17:23:48 +0200
Subject: [Freeswitch-users] Unable to compile mod_sndfile on CentOS 7
Message-ID: <CAMCT=3RJ_5MDoXZdnWdc1D6cf0WUFVdezWuatj4Dcr8x0Fy=cg@mail.gmail.com>

Hey guys,

I'm unable to compile mod_sndfile even though I have installed all the
devel packages.

# yum install libsndfile-devel
Package libsndfile-devel-1.0.25-10.el7.x86_64 already installed and latest
version

# make mod_sndfile
make[1]: Entering directory `/opt/freeswitch'
make  libfreeswitch.la
make[2]: Entering directory `/opt/freeswitch'
make[2]: Leaving directory `/opt/freeswitch'
make[1]: Leaving directory `/opt/freeswitch'
make[1]: Entering directory `/opt/freeswitch/src/mod'
make[2]: Entering directory `/opt/freeswitch/src/mod'

making all mod_sndfile
make[3]: Entering directory `/opt/freeswitch/src/mod/formats/mod_sndfile'
Makefile:888: *** You must install libsndfile-dev to build mod_sndfile.
Stop.
make[3]: Leaving directory `/opt/freeswitch/src/mod/formats/mod_sndfile'
make[2]: *** [mod_sndfile-all] Error 1
make[2]: Leaving directory `/opt/freeswitch/src/mod'
make[1]: *** [mod_sndfile] Error 2
make[1]: Leaving directory `/opt/freeswitch/src/mod'
make: *** [mod_sndfile] Error 2


Any tips?
Best Regards.
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From bobjectsfreeswitch at gmail.com  Mon Feb 29 18:29:42 2016
From: bobjectsfreeswitch at gmail.com (Bob Hartwig)
Date: Mon, 29 Feb 2016 09:29:42 -0600
Subject: [Freeswitch-users] ESL Client --> Multiple FS Servers
In-Reply-To: <CAJUJwtjvsGr0smS-hAHu3KV3Cx3DgpqWZQpM7XVAbU0gSsaVRQ@mail.gmail.com>
References: <CAJUJwtjvsGr0smS-hAHu3KV3Cx3DgpqWZQpM7XVAbU0gSsaVRQ@mail.gmail.com>
Message-ID: <CAAemr3xnruRtY=uACjNc8aOgoyowOWgOsu2Bqo+LR-LtoiVVdA@mail.gmail.com>

Is that inherently more difficult than connecting to one server?  Consider
taking an object-oriented approach:  Make a FreeswitchServer object with
address ivar, have the constructor connect a socket, instantiate once for
each server.



On Mon, Feb 29, 2016 at 8:41 AM, SamyGo <govoiper at gmail.com> wrote:

> Hi All,
>
> I'm looking for some guidance on connecting my ESL client program to
> multiple FS Servers at once. I've written a small code which connects to
> one FS ESL and based on events it generates some reports and also send back
> some actions to FS.
>
> Now I'm looking to expand that form one to multiple FS servers but the
> code should collect all events from every FS and be able to send action to
> any of them whatever the logic be.
>
> Any guides or hints will be much appreciated.
>
> Regards,
> Sammy
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Mon Feb 29 18:53:54 2016
From: abaci64 at gmail.com (Abaci B)
Date: Mon, 29 Feb 2016 10:53:54 -0500
Subject: [Freeswitch-users] Re- End Lua script after HangupHook handled
 without all the extra code to handle the return to the function
In-Reply-To: <MEXPR01MB0151BDA1929ACA8A6E1F8ED8F3BA0@MEXPR01MB0151.ausprd01.prod.outlook.com>
References: <SY3PR01MB015646922642A69251933890F3A70@SY3PR01MB0156.ausprd01.prod.outlook.com>
	<CAFmZFiMFsg5JYGE1Ce6=CVdmiUY5+FUVbm5NHFKE_+enPFNC1g@mail.gmail.com>
	<MEXPR01MB0151BDA1929ACA8A6E1F8ED8F3BA0@MEXPR01MB0151.ausprd01.prod.outlook.com>
Message-ID: <CAFmZFiOKvwcu3VpsYVGqV_GWfGYHxkBY_pHoJ2cKn4mesOghtg@mail.gmail.com>

That's exactly what I noticed, it breaks out of the current function, why
not open a Jira? I don't think it's supposed to behave this way.

On Mon, Feb 29, 2016 at 1:37 AM, Andrew Keil <andrew.keil at visytel.com>
wrote:

> Thanks for your response.
>
>
>
> I have gone through these with no luck.   Like I said the
> session:destroy(???) crashes FreeSWITCH, which is therefore off the list.
> The rest simply interrupt the current function and do no end the script.
>
>
>
> I guess my next move is to see why session:destroy() crashes FreeSWITCH,
> however I am a little snowed under at the moment so if anyone has some time
> to replicate this (only needs one line of code in a Lua script) and pass
> this on to the developers that would be great.
>
>
>
> Andrew
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Abaci B
> *Sent:* Saturday, 27 February 2016 1:29 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Re- End Lua script after HangupHook
> handled without all the extra code to handle the return to the function
>
>
>
> See
> https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:setHangupHook
> for a few ways to exit the lua script (error(), return "exit", return
> "die", s:destroy("error message")). I personally tried return "exit" but it
> seems to me that it only exits the calling function, haven't had a chance
> to look further, it's possible that the calling it from the within a
> function is different. if you play around and figure out please report back.
>
>
>
> On Thu, Feb 25, 2016 at 9:33 PM, Andrew Keil <andrew.keil at visytel.com>
> wrote:
>
> To FreeSWITCH Users,
>
>
>
> See below for a sample template for a Lua Service Script running inside
> FreeSWITCH.
>
>
>
> The issue I have is fairly straightforward.
>
>
>
> I need a function to run when hangup is detected (ie. at the end of the
> call) however I understand this must not delay ending the script.  This
> function is CleanUp().  Then I would like the service to end.
>
>
>
> The problem I am having is if the caller hangs up during the playback of
> ?intro.wav? (as shown inside the MainService() function below), then the
> code jumps to the myHangupHook which calls CleanUp() perfectly, the issue
> is once CleanUp() is complete I would like the Lua script to end there and
> then (ie. at the bottom of CleanUp()).  What actually happens is it returns
> to MainService() and continues to try and play ?info.wav?, unless I either
> check for session:ready() everywhere or add a goto as shown below under
> each streamFile() function call.
>
>
>
> My aim is to reduce extra code and to make the Lua script simpler and
> easier to read.  Also I would like to try and avoid goto statements, which
> I know can be done with if (session:ready()) etc?.
>
>
>
> So is there a way to stop a Lua script running inside FreeSWITCH cleanly?
> I have tried the os.exit() this is barred from use by FreeSWITCH.  I have
> also tried session:destroy() which crashes FreeSWITCH (version 1.6.5 on
> CentOS 6.7, CentOS 7 and windows) 100% of the time!
>
>
>
> I could look further into the Lua additions done by the FreeSWITCH team in
> the source code, however if someone has already solved this then that would
> be the best solution.
>
>
>
> FYI: Obviously the script below is simple, however I am sure that you
> understand if the script was complicated having to use ?*if
> (session:ready()) then ?.?* or ?*if (not session:ready()) then goto
> HANGUPEXIT end?* makes the code ugly.
>
>
>
> Thanks in advance,
>
>
>
> Andrew Keil
>
> *Visytel Pty Ltd*
>
>
>
>
>
>
>
> ------------------------------------------------------------------------------
> Sample Lua Service
> -----------------------------------------------------------------------
>
>
>
> -- Lua template for FreeSWITCH service
>
> -- By: Andrew Keil (Visytel Pty Ltd)
>
> -- Email: support at visytel.com
>
>
>
> -- Setup script wide variables here
>
>
>
> function PreAnswer()
>
>                 freeswitch.consoleLog("INFO", "PRE ANSWER SECTION\n");
>
>                 -- Add your pre answer code from here
>
>
>
>                 -- End of your pre answer code
>
>                 freeswitch.consoleLog("INFO", "PRE ANSWER SECTION
> COMPLETE\n");
>
> end
>
>
>
> function AnswerCaller()
>
>                 session:answer()
>
>                 session:sleep(1000)
>
> end
>
>
>
> function MainService()
>
>                 freeswitch.consoleLog("INFO", "MAIN SERVICE
> SECTION\n");
>
>                 if (session:ready()) then
>
>                                 -- Note (1): If you wish to end the call
> then simply use: goto ENDSERVICE
>
>                                 -- Note (2): To terminate the service
> sooner when HANGUP is detected use: if (not session:ready()) then goto
> HANGUPEXIT end
>
>                                 -- Add your main service code from
> here               (caller would have been answered)
>
>
>
>                                 session:streamFile("intro.wav")
>
>                                 if (not session:ready()) then goto
> HANGUPEXIT end
>
>                                 session:streamFile("info.wav")
>
>                                 if (not session:ready()) then goto
> HANGUPEXIT end
>
>
> session:streamFile("outro.wav")
>
>                                 if (not session:ready()) then goto
> HANGUPEXIT end
>
>
>
>                                 -- End of your main service code
>
>                 end
>
>                 ::ENDSERVICE::
>
>                 if (session:ready()) then
>
>                                 -- End of service so hangup
>
>                                 session:hangup()  -- Should automatically
> jump to CleanUp() via hangup handler if caller still online at this stage
>
>                 end
>
>                 goto END
>
>                 ::HANGUPEXIT::
>
>                 freeswitch.consoleLog("INFO", "END OF SERVICE (HANGUP
> DETECTED)\n");
>
>                 ::END::
>
>                 freeswitch.consoleLog("INFO", "MAIN SERVICE SECTION
> COMPLETE\n");
>
> end
>
>
>
> function CleanUp()
>
>                 freeswitch.consoleLog("INFO", "CLEANUP SECTION\n");
>
>                 -- Add your cleanup code from here (caller would have been
> disconnected)
>
>
>
>                 -- End of your cleanup code
>
>                 freeswitch.consoleLog("INFO", "CLEANUP SECTION
> COMPLETE\n");
>
> end
>
>
>
> function myHangupHook(s, status, arg)
>
>                 session:hangup()
>
>                 CleanUp() -- Run CleanUp function now since the caller has
> disconnected
>
> end
>
>
>
> -- Setup Hangup event handler here
>
> v_hangup = "HANGUP"
>
> session:setHangupHook("myHangupHook", "v_hangup")
>
>
>
> -- Call service functions in order
>
> PreAnswer()
>
> AnswerCaller()
>
> MainService()
>
> -- End of Lua service
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From govoiper at gmail.com  Mon Feb 29 18:55:13 2016
From: govoiper at gmail.com (SamyGo)
Date: Mon, 29 Feb 2016 10:55:13 -0500
Subject: [Freeswitch-users] ESL Client --> Multiple FS Servers
In-Reply-To: <CAAemr3xnruRtY=uACjNc8aOgoyowOWgOsu2Bqo+LR-LtoiVVdA@mail.gmail.com>
References: <CAJUJwtjvsGr0smS-hAHu3KV3Cx3DgpqWZQpM7XVAbU0gSsaVRQ@mail.gmail.com>
	<CAAemr3xnruRtY=uACjNc8aOgoyowOWgOsu2Bqo+LR-LtoiVVdA@mail.gmail.com>
Message-ID: <CAJUJwtjKanreubCR8tUcEcO7tVyhy3=VXN-REgFmKZLvYSzMEA@mail.gmail.com>

Hi Bob,
Thanks for taking out time, I understand what you're saying, and it makes
sense. I have used perl so far and I am considering GoLang for this. (more
of a scripting approach so far)

Can you just tell me this: if it is an ObjOriented code and I connect to
multiple servers at once, the buffer I receive Events into will be common ?
The same loop in which I collect events from all the servers stays the same
too !?

Considering that its been some time I've done some good programming, my dev
skills are decently rusty and hence this email.

Thanks again,
Sammy

On Mon, Feb 29, 2016 at 10:29 AM, Bob Hartwig <bobjectsfreeswitch at gmail.com>
wrote:

> Is that inherently more difficult than connecting to one server?  Consider
> taking an object-oriented approach:  Make a FreeswitchServer object with
> address ivar, have the constructor connect a socket, instantiate once for
> each server.
>
>
>
> On Mon, Feb 29, 2016 at 8:41 AM, SamyGo <govoiper at gmail.com> wrote:
>
>> Hi All,
>>
>> I'm looking for some guidance on connecting my ESL client program to
>> multiple FS Servers at once. I've written a small code which connects to
>> one FS ESL and based on events it generates some reports and also send back
>> some actions to FS.
>>
>> Now I'm looking to expand that form one to multiple FS servers but the
>> code should collect all events from every FS and be able to send action to
>> any of them whatever the logic be.
>>
>> Any guides or hints will be much appreciated.
>>
>> Regards,
>> Sammy
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Mon Feb 29 19:15:51 2016
From: abaci64 at gmail.com (Abaci B)
Date: Mon, 29 Feb 2016 11:15:51 -0500
Subject: [Freeswitch-users] Re- End Lua script after HangupHook handled
 without all the extra code to handle the return to the function
In-Reply-To: <CAFmZFiOKvwcu3VpsYVGqV_GWfGYHxkBY_pHoJ2cKn4mesOghtg@mail.gmail.com>
References: <SY3PR01MB015646922642A69251933890F3A70@SY3PR01MB0156.ausprd01.prod.outlook.com>
	<CAFmZFiMFsg5JYGE1Ce6=CVdmiUY5+FUVbm5NHFKE_+enPFNC1g@mail.gmail.com>
	<MEXPR01MB0151BDA1929ACA8A6E1F8ED8F3BA0@MEXPR01MB0151.ausprd01.prod.outlook.com>
	<CAFmZFiOKvwcu3VpsYVGqV_GWfGYHxkBY_pHoJ2cKn4mesOghtg@mail.gmail.com>
Message-ID: <CAFmZFiN1MEJHcoZ90-sUJqtXwb6LHO_iEohD_S=3BX6GVpweKA@mail.gmail.com>

I don't see in your example above where you have a return "exit" in your
hangup hook function
see https://freeswitch.org/jira/browse/FS-3841
seems like the "exit" or "die" needs to be returned directly from the
hanguphook function so try from that function (not cleanup function) to
return "exit" or "dye", also it seems

On Mon, Feb 29, 2016 at 10:53 AM, Abaci B <abaci64 at gmail.com> wrote:

> That's exactly what I noticed, it breaks out of the current function, why
> not open a Jira? I don't think it's supposed to behave this way.
>
> On Mon, Feb 29, 2016 at 1:37 AM, Andrew Keil <andrew.keil at visytel.com>
> wrote:
>
>> Thanks for your response.
>>
>>
>>
>> I have gone through these with no luck.   Like I said the
>> session:destroy(???) crashes FreeSWITCH, which is therefore off the list.
>> The rest simply interrupt the current function and do no end the script.
>>
>>
>>
>> I guess my next move is to see why session:destroy() crashes FreeSWITCH,
>> however I am a little snowed under at the moment so if anyone has some time
>> to replicate this (only needs one line of code in a Lua script) and pass
>> this on to the developers that would be great.
>>
>>
>>
>> Andrew
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Abaci B
>> *Sent:* Saturday, 27 February 2016 1:29 AM
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* Re: [Freeswitch-users] Re- End Lua script after HangupHook
>> handled without all the extra code to handle the return to the function
>>
>>
>>
>> See
>> https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:setHangupHook
>> for a few ways to exit the lua script (error(), return "exit", return
>> "die", s:destroy("error message")). I personally tried return "exit" but it
>> seems to me that it only exits the calling function, haven't had a chance
>> to look further, it's possible that the calling it from the within a
>> function is different. if you play around and figure out please report back.
>>
>>
>>
>> On Thu, Feb 25, 2016 at 9:33 PM, Andrew Keil <andrew.keil at visytel.com>
>> wrote:
>>
>> To FreeSWITCH Users,
>>
>>
>>
>> See below for a sample template for a Lua Service Script running inside
>> FreeSWITCH.
>>
>>
>>
>> The issue I have is fairly straightforward.
>>
>>
>>
>> I need a function to run when hangup is detected (ie. at the end of the
>> call) however I understand this must not delay ending the script.  This
>> function is CleanUp().  Then I would like the service to end.
>>
>>
>>
>> The problem I am having is if the caller hangs up during the playback of
>> ?intro.wav? (as shown inside the MainService() function below), then the
>> code jumps to the myHangupHook which calls CleanUp() perfectly, the issue
>> is once CleanUp() is complete I would like the Lua script to end there and
>> then (ie. at the bottom of CleanUp()).  What actually happens is it returns
>> to MainService() and continues to try and play ?info.wav?, unless I either
>> check for session:ready() everywhere or add a goto as shown below under
>> each streamFile() function call.
>>
>>
>>
>> My aim is to reduce extra code and to make the Lua script simpler and
>> easier to read.  Also I would like to try and avoid goto statements, which
>> I know can be done with if (session:ready()) etc?.
>>
>>
>>
>> So is there a way to stop a Lua script running inside FreeSWITCH
>> cleanly?  I have tried the os.exit() this is barred from use by
>> FreeSWITCH.  I have also tried session:destroy() which crashes FreeSWITCH
>> (version 1.6.5 on CentOS 6.7, CentOS 7 and windows) 100% of the time!
>>
>>
>>
>> I could look further into the Lua additions done by the FreeSWITCH team
>> in the source code, however if someone has already solved this then that
>> would be the best solution.
>>
>>
>>
>> FYI: Obviously the script below is simple, however I am sure that you
>> understand if the script was complicated having to use ?*if
>> (session:ready()) then ?.?* or ?*if (not session:ready()) then goto
>> HANGUPEXIT end?* makes the code ugly.
>>
>>
>>
>> Thanks in advance,
>>
>>
>>
>> Andrew Keil
>>
>> *Visytel Pty Ltd*
>>
>>
>>
>>
>>
>>
>>
>> ------------------------------------------------------------------------------
>> Sample Lua Service
>> -----------------------------------------------------------------------
>>
>>
>>
>> -- Lua template for FreeSWITCH service
>>
>> -- By: Andrew Keil (Visytel Pty Ltd)
>>
>> -- Email: support at visytel.com
>>
>>
>>
>> -- Setup script wide variables here
>>
>>
>>
>> function PreAnswer()
>>
>>                 freeswitch.consoleLog("INFO", "PRE ANSWER SECTION\n");
>>
>>                 -- Add your pre answer code from here
>>
>>
>>
>>                 -- End of your pre answer code
>>
>>                 freeswitch.consoleLog("INFO", "PRE ANSWER SECTION
>> COMPLETE\n");
>>
>> end
>>
>>
>>
>> function AnswerCaller()
>>
>>                 session:answer()
>>
>>                 session:sleep(1000)
>>
>> end
>>
>>
>>
>> function MainService()
>>
>>                 freeswitch.consoleLog("INFO", "MAIN SERVICE
>> SECTION\n");
>>
>>                 if (session:ready()) then
>>
>>                                 -- Note (1): If you wish to end the call
>> then simply use: goto ENDSERVICE
>>
>>                                 -- Note (2): To terminate the service
>> sooner when HANGUP is detected use: if (not session:ready()) then goto
>> HANGUPEXIT end
>>
>>                                 -- Add your main service code from
>> here               (caller would have been answered)
>>
>>
>>
>>                                 session:streamFile("intro.wav")
>>
>>                                 if (not session:ready()) then goto
>> HANGUPEXIT end
>>
>>                                 session:streamFile("info.wav")
>>
>>                                 if (not session:ready()) then goto
>> HANGUPEXIT end
>>
>>
>> session:streamFile("outro.wav")
>>
>>                                 if (not session:ready()) then goto
>> HANGUPEXIT end
>>
>>
>>
>>                                 -- End of your main service code
>>
>>                 end
>>
>>                 ::ENDSERVICE::
>>
>>                 if (session:ready()) then
>>
>>                                 -- End of service so hangup
>>
>>                                 session:hangup()  -- Should automatically
>> jump to CleanUp() via hangup handler if caller still online at this stage
>>
>>                 end
>>
>>                 goto END
>>
>>                 ::HANGUPEXIT::
>>
>>                 freeswitch.consoleLog("INFO", "END OF SERVICE (HANGUP
>> DETECTED)\n");
>>
>>                 ::END::
>>
>>                 freeswitch.consoleLog("INFO", "MAIN SERVICE SECTION
>> COMPLETE\n");
>>
>> end
>>
>>
>>
>> function CleanUp()
>>
>>                 freeswitch.consoleLog("INFO", "CLEANUP SECTION\n");
>>
>>                 -- Add your cleanup code from here (caller would have
>> been disconnected)
>>
>>
>>
>>                 -- End of your cleanup code
>>
>>                 freeswitch.consoleLog("INFO", "CLEANUP SECTION
>> COMPLETE\n");
>>
>> end
>>
>>
>>
>> function myHangupHook(s, status, arg)
>>
>>                 session:hangup()
>>
>>                 CleanUp() -- Run CleanUp function now since the caller
>> has disconnected
>>
>> end
>>
>>
>>
>> -- Setup Hangup event handler here
>>
>> v_hangup = "HANGUP"
>>
>> session:setHangupHook("myHangupHook", "v_hangup")
>>
>>
>>
>> -- Call service functions in order
>>
>> PreAnswer()
>>
>> AnswerCaller()
>>
>> MainService()
>>
>> -- End of Lua service
>>
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From bobjectsfreeswitch at gmail.com  Mon Feb 29 19:34:35 2016
From: bobjectsfreeswitch at gmail.com (Bob Hartwig)
Date: Mon, 29 Feb 2016 10:34:35 -0600
Subject: [Freeswitch-users] ESL Client --> Multiple FS Servers
In-Reply-To: <CAJUJwtjKanreubCR8tUcEcO7tVyhy3=VXN-REgFmKZLvYSzMEA@mail.gmail.com>
References: <CAJUJwtjvsGr0smS-hAHu3KV3Cx3DgpqWZQpM7XVAbU0gSsaVRQ@mail.gmail.com>
	<CAAemr3xnruRtY=uACjNc8aOgoyowOWgOsu2Bqo+LR-LtoiVVdA@mail.gmail.com>
	<CAJUJwtjKanreubCR8tUcEcO7tVyhy3=VXN-REgFmKZLvYSzMEA@mail.gmail.com>
Message-ID: <CAAemr3wJUxPKUsKRESKSZxg5hrU3MHfQYJMMjqavU_yk2wecrg@mail.gmail.com>

I guess it depends on whether you need to send commands in reply to
events.  If so, you'll probably want to have separate queues and servicing
threads for each server.  Your requirements can probably tell you better
than I can.

Bob



On Mon, Feb 29, 2016 at 9:55 AM, SamyGo <govoiper at gmail.com> wrote:

> Hi Bob,
> Thanks for taking out time, I understand what you're saying, and it makes
> sense. I have used perl so far and I am considering GoLang for this. (more
> of a scripting approach so far)
>
> Can you just tell me this: if it is an ObjOriented code and I connect to
> multiple servers at once, the buffer I receive Events into will be common ?
> The same loop in which I collect events from all the servers stays the same
> too !?
>
> Considering that its been some time I've done some good programming, my
> dev skills are decently rusty and hence this email.
>
> Thanks again,
> Sammy
>
> On Mon, Feb 29, 2016 at 10:29 AM, Bob Hartwig <
> bobjectsfreeswitch at gmail.com> wrote:
>
>> Is that inherently more difficult than connecting to one server?
>> Consider taking an object-oriented approach:  Make a FreeswitchServer
>> object with address ivar, have the constructor connect a socket,
>> instantiate once for each server.
>>
>>
>>
>> On Mon, Feb 29, 2016 at 8:41 AM, SamyGo <govoiper at gmail.com> wrote:
>>
>>> Hi All,
>>>
>>> I'm looking for some guidance on connecting my ESL client program to
>>> multiple FS Servers at once. I've written a small code which connects to
>>> one FS ESL and based on events it generates some reports and also send back
>>> some actions to FS.
>>>
>>> Now I'm looking to expand that form one to multiple FS servers but the
>>> code should collect all events from every FS and be able to send action to
>>> any of them whatever the logic be.
>>>
>>> Any guides or hints will be much appreciated.
>>>
>>> Regards,
>>> Sammy
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Shawn.Wheeler at interlockconcepts.com  Mon Feb 29 08:18:17 2016
From: Shawn.Wheeler at interlockconcepts.com (Shawn Wheeler)
Date: Mon, 29 Feb 2016 05:18:17 +0000
Subject: [Freeswitch-users] Question on FreeSWITCH-1.7.0-0a024c4ecb-64bit
	and service startup
Message-ID: <CY1PR0201MB1497E65E7C861795E0D6EF1DE0BA0@CY1PR0201MB1497.namprd02.prod.outlook.com>

Parameters
Virtual Machine running Windows Server 2012 R2 base machine.
I have installed FreeSWITCH-1.7.0-0a024c4ecb-64bit

I have run into two issues as noted below.  Any help would be most appreciated.

Item 1
When I tried to start the service I received the following error.

?Windows could not start the FreeSWITCH Multi Protocol Switch service on Local Computer
Error 1053:  The service did not respond to start or control request in a timely fashion?

Item 2
I also received this error when trying to start the CLI

?The program can't stat because VCRUNTIME140.dll is missing from your computer.
Try reinstalling the program to fix this problem.?

I assume it was due to the service not starting.

Thank you all in advance.

Shawn

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From delvin.friends at gmail.com  Mon Feb 29 21:46:38 2016
From: delvin.friends at gmail.com (Delvin Varghese)
Date: Mon, 29 Feb 2016 18:46:38 +0000
Subject: [Freeswitch-users] Recording each leg of a conference call
	separately
Message-ID: <4567C63B-8AF4-40A4-B9F5-915E4073F1CF@gmail.com>

Hi all,

I am familiar with the record and record_session commands in freeswitch to initiate recordings, and also using ESL.. But is there a way to record ONLY the speech from one device at a time in separate recordings? The way i am envisioning is that if there are 9 different devices involved in a conference call, that at the end of it I would be able to access 9 recordings (each containing only the outbound voice from that device)?

Is this possible? What are people?s thoughts?

Delvin



From yehavi.bourvine at gmail.com  Mon Feb 29 21:55:27 2016
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Mon, 29 Feb 2016 20:55:27 +0200
Subject: [Freeswitch-users] Unable to compile mod_sndfile on CentOS 7
In-Reply-To: <CAMCT=3RJ_5MDoXZdnWdc1D6cf0WUFVdezWuatj4Dcr8x0Fy=cg@mail.gmail.com>
References: <CAMCT=3RJ_5MDoXZdnWdc1D6cf0WUFVdezWuatj4Dcr8x0Fy=cg@mail.gmail.com>
Message-ID: <CAAtWKeFFYFgc+5Cru=ERWEgHmZWTVCPWNL8PhcoNGMO6vHV3rg@mail.gmail.com>

If you do not need this module then simply comment it out in modules.conf;
that's what I did...

                    __Yehavi:

2016-02-29 17:23 GMT+02:00 Vladislav Ivanov <deforceczt at gmail.com>:

> Hey guys,
>
> I'm unable to compile mod_sndfile even though I have installed all the
> devel packages.
>
> # yum install libsndfile-devel
> Package libsndfile-devel-1.0.25-10.el7.x86_64 already installed and latest
> version
>
> # make mod_sndfile
> make[1]: Entering directory `/opt/freeswitch'
> make  libfreeswitch.la
> make[2]: Entering directory `/opt/freeswitch'
> make[2]: Leaving directory `/opt/freeswitch'
> make[1]: Leaving directory `/opt/freeswitch'
> make[1]: Entering directory `/opt/freeswitch/src/mod'
> make[2]: Entering directory `/opt/freeswitch/src/mod'
>
> making all mod_sndfile
> make[3]: Entering directory `/opt/freeswitch/src/mod/formats/mod_sndfile'
> Makefile:888: *** You must install libsndfile-dev to build mod_sndfile.
> Stop.
> make[3]: Leaving directory `/opt/freeswitch/src/mod/formats/mod_sndfile'
> make[2]: *** [mod_sndfile-all] Error 1
> make[2]: Leaving directory `/opt/freeswitch/src/mod'
> make[1]: *** [mod_sndfile] Error 2
> make[1]: Leaving directory `/opt/freeswitch/src/mod'
> make: *** [mod_sndfile] Error 2
>
>
> Any tips?
> 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 krice at freeswitch.org  Mon Feb 29 21:57:15 2016
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 29 Feb 2016 12:57:15 -0600
Subject: [Freeswitch-users] Question on
	FreeSWITCH-1.7.0-0a024c4ecb-64bit	and service startup
In-Reply-To: <CY1PR0201MB1497E65E7C861795E0D6EF1DE0BA0@CY1PR0201MB1497.namprd02.prod.outlook.com>
References: <CY1PR0201MB1497E65E7C861795E0D6EF1DE0BA0@CY1PR0201MB1497.namprd02.prod.outlook.com>
Message-ID: <42c401d17323$024793f0$06d6bbd0$@freeswitch.org>

Just install the VC runtime from microsoft

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shawn Wheeler
Sent: Sunday, February 28, 2016 11:18 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Question on FreeSWITCH-1.7.0-0a024c4ecb-64bit and service startup

 

Parameters

Virtual Machine running Windows Server 2012 R2 base machine.

I have installed FreeSWITCH-1.7.0-0a024c4ecb-64bit

 

I have run into two issues as noted below.  Any help would be most appreciated.

 

Item 1

When I tried to start the service I received the following error.

 

?Windows could not start the FreeSWITCH Multi Protocol Switch service on Local Computer

Error 1053:  The service did not respond to start or control request in a timely fashion?

 

Item 2

I also received this error when trying to start the CLI

 

?The program can't stat because VCRUNTIME140.dll is missing from your computer.  

Try reinstalling the program to fix this problem.?

 

I assume it was due to the service not starting. 

 

Thank you all in advance.

 

Shawn

 

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From ynasida at gmail.com  Mon Feb 29 22:01:25 2016
From: ynasida at gmail.com (=?UTF-8?B?0K7RgNC40Lkg0J3QsNGB0LjQtNCw?=)
Date: Mon, 29 Feb 2016 22:01:25 +0300
Subject: [Freeswitch-users] segfaults because odbc mysql driver
Message-ID: <CADihe3J-rWXxtP3=ycwTPUMU+TUb=v53w=L=NakwqMhZ+Fo+Uw@mail.gmail.com>

Hi guys,

As far as I know there is one already known problems with FS segfaults that
 happen because of odbc mysql driver.
I have latest recommended setup with debian 8.3 + FS 1.6.
Somebody mentioned that version of unixODBC should be higher than 2.3 to
solve this but... I have  unixODBC 2.3.1 and it doesn't help.

Everybody use pgsql ? :)

Please advice if somebody  use mysql with FS 1.6 and still happy.

I also have backtrace file here - https://freeswitch.org/jira/browse/FS-8857

and will be glad any ideas.

Thanks.
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From deforceczt at gmail.com  Mon Feb 29 22:05:45 2016
From: deforceczt at gmail.com (Vladislav Ivanov)
Date: Mon, 29 Feb 2016 21:05:45 +0200
Subject: [Freeswitch-users] Unable to compile mod_sndfile on CentOS 7
In-Reply-To: <CAAtWKeFFYFgc+5Cru=ERWEgHmZWTVCPWNL8PhcoNGMO6vHV3rg@mail.gmail.com>
References: <CAMCT=3RJ_5MDoXZdnWdc1D6cf0WUFVdezWuatj4Dcr8x0Fy=cg@mail.gmail.com>
	<CAAtWKeFFYFgc+5Cru=ERWEgHmZWTVCPWNL8PhcoNGMO6vHV3rg@mail.gmail.com>
Message-ID: <CAMCT=3SmJTF0Z5C0PTkLx=fsBXPFeixmPB3xN071f-KyFE+koQ@mail.gmail.com>

Yeah, I need it...

2016-02-29 20:55 GMT+02:00 Yehavi Bourvine <yehavi.bourvine at gmail.com>:

> If you do not need this module then simply comment it out in modules.conf;
> that's what I did...
>
>                     __Yehavi:
>
> 2016-02-29 17:23 GMT+02:00 Vladislav Ivanov <deforceczt at gmail.com>:
>
>> Hey guys,
>>
>> I'm unable to compile mod_sndfile even though I have installed all the
>> devel packages.
>>
>> # yum install libsndfile-devel
>> Package libsndfile-devel-1.0.25-10.el7.x86_64 already installed and
>> latest version
>>
>> # make mod_sndfile
>> make[1]: Entering directory `/opt/freeswitch'
>> make  libfreeswitch.la
>> make[2]: Entering directory `/opt/freeswitch'
>> make[2]: Leaving directory `/opt/freeswitch'
>> make[1]: Leaving directory `/opt/freeswitch'
>> make[1]: Entering directory `/opt/freeswitch/src/mod'
>> make[2]: Entering directory `/opt/freeswitch/src/mod'
>>
>> making all mod_sndfile
>> make[3]: Entering directory `/opt/freeswitch/src/mod/formats/mod_sndfile'
>> Makefile:888: *** You must install libsndfile-dev to build mod_sndfile.
>> Stop.
>> make[3]: Leaving directory `/opt/freeswitch/src/mod/formats/mod_sndfile'
>> make[2]: *** [mod_sndfile-all] Error 1
>> make[2]: Leaving directory `/opt/freeswitch/src/mod'
>> make[1]: *** [mod_sndfile] Error 2
>> make[1]: Leaving directory `/opt/freeswitch/src/mod'
>> make: *** [mod_sndfile] Error 2
>>
>>
>> Any tips?
>> 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 govoiper at gmail.com  Mon Feb 29 22:09:51 2016
From: govoiper at gmail.com (SamyGo)
Date: Mon, 29 Feb 2016 14:09:51 -0500
Subject: [Freeswitch-users] Unable to compile mod_sndfile on CentOS 7
In-Reply-To: <CAMCT=3SmJTF0Z5C0PTkLx=fsBXPFeixmPB3xN071f-KyFE+koQ@mail.gmail.com>
References: <CAMCT=3RJ_5MDoXZdnWdc1D6cf0WUFVdezWuatj4Dcr8x0Fy=cg@mail.gmail.com>
	<CAAtWKeFFYFgc+5Cru=ERWEgHmZWTVCPWNL8PhcoNGMO6vHV3rg@mail.gmail.com>
	<CAMCT=3SmJTF0Z5C0PTkLx=fsBXPFeixmPB3xN071f-KyFE+koQ@mail.gmail.com>
Message-ID: <CAJUJwths++ghGMS=9k6LsnC1m6PTaUKDViFMHqTorRqZzF69=A@mail.gmail.com>

Hi,
I recently got into trouble where I compiled FS on ubuntu 14.04 and got
error on this module - so I commented it out. Turned out that I was unable
to play .wav files in FS.
Hence I had to compile this module using an older version of libsndfile-dev
since Ubuntu 14.04 installed libsndfile1.dev.

It seems like you already have the same version which worked for me, maybe
try executing command "ldconfig" on shell and then try.

Thanks
Sammy

On Mon, Feb 29, 2016 at 2:05 PM, Vladislav Ivanov <deforceczt at gmail.com>
wrote:

> Yeah, I need it...
>
> 2016-02-29 20:55 GMT+02:00 Yehavi Bourvine <yehavi.bourvine at gmail.com>:
>
>> If you do not need this module then simply comment it out in
>> modules.conf; that's what I did...
>>
>>                     __Yehavi:
>>
>> 2016-02-29 17:23 GMT+02:00 Vladislav Ivanov <deforceczt at gmail.com>:
>>
>>> Hey guys,
>>>
>>> I'm unable to compile mod_sndfile even though I have installed all the
>>> devel packages.
>>>
>>> # yum install libsndfile-devel
>>> Package libsndfile-devel-1.0.25-10.el7.x86_64 already installed and
>>> latest version
>>>
>>> # make mod_sndfile
>>> make[1]: Entering directory `/opt/freeswitch'
>>> make  libfreeswitch.la
>>> make[2]: Entering directory `/opt/freeswitch'
>>> make[2]: Leaving directory `/opt/freeswitch'
>>> make[1]: Leaving directory `/opt/freeswitch'
>>> make[1]: Entering directory `/opt/freeswitch/src/mod'
>>> make[2]: Entering directory `/opt/freeswitch/src/mod'
>>>
>>> making all mod_sndfile
>>> make[3]: Entering directory `/opt/freeswitch/src/mod/formats/mod_sndfile'
>>> Makefile:888: *** You must install libsndfile-dev to build mod_sndfile.
>>> Stop.
>>> make[3]: Leaving directory `/opt/freeswitch/src/mod/formats/mod_sndfile'
>>> make[2]: *** [mod_sndfile-all] Error 1
>>> make[2]: Leaving directory `/opt/freeswitch/src/mod'
>>> make[1]: *** [mod_sndfile] Error 2
>>> make[1]: Leaving directory `/opt/freeswitch/src/mod'
>>> make: *** [mod_sndfile] Error 2
>>>
>>>
>>> Any tips?
>>> 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 jurij.ivo at gmail.com  Mon Feb 29 22:27:30 2016
From: jurij.ivo at gmail.com (Jurijs Ivolga)
Date: Mon, 29 Feb 2016 21:27:30 +0200
Subject: [Freeswitch-users] Call dropping after 32 seconds
In-Reply-To: <CAPQet1xzzXKUk=4jJRqtZ4rwtLUDEtr7n-eZBGvQp6kvinb+aw@mail.gmail.com>
References: <CAPQet1zs08bGNnGDVmsLH3R3VbvMCJW=8Cx3U+KPxwGpYSuzAA@mail.gmail.com>
	<0000015307b150a5-132a41d6-5bea-4af1-9686-2653df68a3b6-000000@email.amazonses.com>
	<CAPQet1wJPNZj7S4ezzd=OSDqHzLKQMieOWvtL8Qm5ht1PcHyBw@mail.gmail.com>
	<CAGYpU64NgPRKPCywdNn3whBh8VH6=eYLy=PJPGrYn8JJ5iMPGQ@mail.gmail.com>
	<CAPQet1xzzXKUk=4jJRqtZ4rwtLUDEtr7n-eZBGvQp6kvinb+aw@mail.gmail.com>
Message-ID: <CAGYpU642mfmHj1g3kUHxy0xeLOd2nvh7yfDrf4VErhrbOeFwoQ@mail.gmail.com>

Hi Rutu,

Firewall should not be an issue. If you think that it may change SIP
packets, you can always use TLS. I doubt that it is good idea to add one
more server in set-up, just because of firewall, you just need to configure
all of your servers, devices properly.

If you can, you should eliminate unnecessary servers from your set-up.

>From what you described before, it might be issue not connected to NAT, but
because Opensips wasn't configured properly. I had similar issue when
Kamailio(Opensip is fork from Kamailio project, so they almost identical)
was wrongly configured, particularly path header wasn't inserted by
Kamailio. But this 30 seconds timeout is quite often NAT issue, but again
if you have NAT issue, you should not blame FW or anything else, you should
just configure your Freeswitch and Opensips properly. Almost all devices in
internet are behind NAT and almost all of them works perfectly with VoIP.

So if you need, help, then please send full sip trace, so I can take a look
on it.

With kind regards,

Jurijs

2016-02-29 15:15 GMT+02:00 Rutu Patel <rutu.patel at inextrix.com>:

> Hi Jurijs,
>
> We have to consider the setup with freeswitch and opensips only.
> registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips
> server)
>
> And there is a firewall and we have added x.x.x.3 IP there.
>
> Is it possible there is some network related issue Or any sip profile
> parameter we need to set?
>
> --
> Thanks,
> Rutu Patel
>
> On Tue, Feb 23, 2016 at 1:07 PM, Jurijs Ivolga <jurij.ivo at gmail.com>
> wrote:
>
>> Hi,
>>
>> 1) You have very complex set-up and I doubt that you need it.
>>
>> 2) As far as you have user with ip x.x.x.174 and opensips server with
>> same ip x.x.x.174 it very hard to debug. So I propose you to send new log
>> where will be difference between user ip and opensips IP.
>>
>> 3) If you have possibility, try to register directly with a user to
>> x.x.x.3 gateway and check if same issue still exists, if there is no such
>> issue anymore, then thee is definitely issue in your opensips x.x.x.174 and
>> freeswitch x.x.x.166. My point here is that you need to isolate issue and
>> to understand what part of your set-up works as expected and what is faulty.
>>
>> With kind regards,
>>
>> Jurijs
>>
>> 2016-02-23 7:51 GMT+02:00 Rutu Patel <rutu.patel at inextrix.com>:
>>
>>> Thanks for the reply.
>>>
>>> Got your point about NATing issue and no response of 200 OK and as a
>>> resoult ACK Timeout.
>>> So, now to resolve the issue, if you can assist, what could be the
>>> possible fixies?
>>> From where can i start? where to look?
>>>
>>> Thanks.
>>>
>>> --
>>> Thanks,
>>> Rutu Patel
>>>
>>> <http://www.inextrix.com>
>>>
>>> On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>>
>>>> 5 second response: 32 seconds is a timer/[network/NAT] issue.
>>>>
>>>> You have lots of 200s to the user since it's waiting for an ACK and
>>>> keeps retrying, but for whatever network reason (router... sip alg?), it
>>>> isn't getting one, so it triggers a timer to stop the call.
>>>>
>>>>
>>>> -Avi Marcus
>>>> BestFone
>>>>
>>>> On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel <rutu.patel at inextrix.com>
>>>> wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>> Having issue of call dropping after 32 seconds, here are the details-
>>>>>
>>>>> x.x.x.174: opensips server
>>>>> x.x.x.166: freeswitch server
>>>>> x.x.x.3:     another opensips server which is registered as gateway on
>>>>> above freeswitch server
>>>>> x.x.x.6: freeswitch server
>>>>> x.x.x.47:  server through which the user is registered
>>>>> I am trying to call from xxxx9 to xxxxxxx29858
>>>>> xxxxxxx00181 is caller-id name and caller-id number
>>>>>
>>>>> Call flow is like this:
>>>>> registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174
>>>>> (opensips server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server)
>>>>>
>>>>> Outbound-proxy is set to x.x.x.174 in Gateway configuration.
>>>>>
>>>>> 1) Call is initiated by the user(xxxx9) registered with host x.x.x.174
>>>>> 2) Call hit the freeswitch server x.x.x.166
>>>>> 3) After '180 Ringing' and '183 Session Progress' packet
>>>>> sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the
>>>>> gateway x.x.x.3
>>>>> But after 32 seconds call is dropped,
>>>>> Within 32 seconds audio is ok from both end so it should not be the
>>>>> RTP issue.
>>>>> Here I have attached the file with sip logs, you can observer from the
>>>>> file that, there are many '200 OK' from x.x.x.174 to x.x.x.166
>>>>> and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then
>>>>> 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped.
>>>>>
>>>>> What is wrong here? Any help would be appreciated here.
>>>>>
>>>>> Here is the file with sip logs
>>>>> --
>>>>> Thanks,
>>>>> Rutu Patel
>>>>> <http://www.inextrix.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
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> Jurijs
>>
>> 2016-02-23 7:51 GMT+02:00 Rutu Patel <rutu.patel at inextrix.com>:
>>
>>> Thanks for the reply.
>>>
>>> Got your point about NATing issue and no response of 200 OK and as a
>>> resoult ACK Timeout.
>>> So, now to resolve the issue, if you can assist, what could be the
>>> possible fixies?
>>> From where can i start? where to look?
>>>
>>> Thanks.
>>>
>>> --
>>> Thanks,
>>> Rutu Patel
>>>
>>> <http://www.inextrix.com>
>>>
>>> On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>>
>>>> 5 second response: 32 seconds is a timer/[network/NAT] issue.
>>>>
>>>> You have lots of 200s to the user since it's waiting for an ACK and
>>>> keeps retrying, but for whatever network reason (router... sip alg?), it
>>>> isn't getting one, so it triggers a timer to stop the call.
>>>>
>>>>
>>>> -Avi Marcus
>>>> BestFone
>>>>
>>>> On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel <rutu.patel at inextrix.com>
>>>> wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>> Having issue of call dropping after 32 seconds, here are the details-
>>>>>
>>>>> x.x.x.174: opensips server
>>>>> x.x.x.166: freeswitch server
>>>>> x.x.x.3:     another opensips server which is registered as gateway on
>>>>> above freeswitch server
>>>>> x.x.x.6: freeswitch server
>>>>> x.x.x.47:  server through which the user is registered
>>>>> I am trying to call from xxxx9 to xxxxxxx29858
>>>>> xxxxxxx00181 is caller-id name and caller-id number
>>>>>
>>>>> Call flow is like this:
>>>>> registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174
>>>>> (opensips server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server)
>>>>>
>>>>> Outbound-proxy is set to x.x.x.174 in Gateway configuration.
>>>>>
>>>>> 1) Call is initiated by the user(xxxx9) registered with host x.x.x.174
>>>>> 2) Call hit the freeswitch server x.x.x.166
>>>>> 3) After '180 Ringing' and '183 Session Progress' packet
>>>>> sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the
>>>>> gateway x.x.x.3
>>>>> But after 32 seconds call is dropped,
>>>>> Within 32 seconds audio is ok from both end so it should not be the
>>>>> RTP issue.
>>>>> Here I have attached the file with sip logs, you can observer from the
>>>>> file that, there are many '200 OK' from x.x.x.174 to x.x.x.166
>>>>> and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then
>>>>> 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped.
>>>>>
>>>>> What is wrong here? Any help would be appreciated here.
>>>>>
>>>>> Here is the file with sip logs
>>>>> --
>>>>> Thanks,
>>>>> Rutu Patel
>>>>> <http://www.inextrix.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
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 ynasida at gmail.com  Mon Feb 29 23:00:27 2016
From: ynasida at gmail.com (=?UTF-8?B?0K7RgNC40Lkg0J3QsNGB0LjQtNCw?=)
Date: Mon, 29 Feb 2016 23:00:27 +0300
Subject: [Freeswitch-users] where mod_com_g729 ?
In-Reply-To: <CAEJMVkCASCL46CYtuo0gH_gFDrV2bUCA3Tnd+qUs1Vz40PWvAA@mail.gmail.com>
References: <CADihe3J3kv2RJ8DeueusW6aAkzGWNxR2h5zaKiLKV9asJg6LVg@mail.gmail.com>
	<CABMX_w-ga8GM9s+JHyWjSOAMLRqfJ=o5+YTLYww5V8qNbD7A4Q@mail.gmail.com>
	<CADihe3JFLUtzLPQk6J+0rsVtVP5FG88uhbjyNeBOfhqP1brWfg@mail.gmail.com>
	<DE690397-43CB-420F-BAC5-5E7DA57B81E3@freeswitch.org>
	<CADihe3+3xCRQXtoKJsy4=Oc6XaZXHCYEqfxME8na_Oh=s_TXhA@mail.gmail.com>
	<CAEJMVkCASCL46CYtuo0gH_gFDrV2bUCA3Tnd+qUs1Vz40PWvAA@mail.gmail.com>
Message-ID: <CADihe3+j=6WQwsw_Cbo4jFW+wxj=wQRwLyK4fTHmsVrGydEwyw@mail.gmail.com>

no, I installed all things using apt-get and did not set nonstandard path
anywhere.
29 ??? 2016 ?. 17:31 ???????????? "Brian West" <brian at freeswitch.org>
???????:

> Are you using a nonstandard install path?
>
> On Monday, February 29, 2016, ???? ?????? <ynasida at gmail.com> wrote:
>
>> Thanks guys.
>> I just had to set manually the path for  script fs-latest-installer-v1.6
>> <bin_path> <mods_path> <config_path>
>> Otherwise mod_com_g729.so was not installed anywhere (without any warning
>> message).
>>
>> 2016-02-28 0:00 GMT+03:00 Ken Rice <krice at freeswitch.org>:
>>
>>> I'm pretty sure the link he gave you is the g729 commercial mod
>>> installation instructions and answers your question
>>>
>>> Sent from my iPhone
>>>
>>> On Feb 27, 2016, at 12:57 PM, ???? ?????? <ynasida at gmail.com> wrote:
>>>
>>> Looks like you didn't read my question.
>>>
>>> 2016-02-27 18:49 GMT+03:00 ?talo Rossi <italo at freeswitch.org>:
>>>
>>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_com_g729
>>>>
>>>> On Sat, Feb 27, 2016 at 5:36 AM, ???? ?????? <ynasida at gmail.com> wrote:
>>>>
>>>>> Hi list
>>>>>
>>>>> I have installed FS 1.6.6 via apt-get method (all modules) and don't
>>>>> see mod_com_g729 anywhere.
>>>>>
>>>>> Should I install it from source ?
>>>>>
>>>>> Please advice.
>>>>>
>>>>> # apt-cache search freeswitch | grep 729
>>>>> freeswitch-mod-g729 - mod_g729 for FreeSWITCH
>>>>> freeswitch-mod-g729-dbg - mod_g729 for FreeSWITCH (debug)
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
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>>>>> 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
>>>>
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>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
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>>> http://www.cluecon.com
>>>
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>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting 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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>>> http://www.freeswitch.org
>>>
>>
>>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
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>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
>
> _________________________________________________________________________
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> consulting at freeswitch.org
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>
> Official FreeSWITCH Sites
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> http://confluence.freeswitch.org
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>
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