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 E